Spip Ssip VVX Admin Guide SIP 3 2 2 Eng

Administrator’s Guide for the
Polycom® SoundPoint®
IP/SoundStation® IP/ VVX™
Family
SIP 3.2.2 | November 2009 | 1725-11530-322 Rev. A

Trademark Information
POLYCOM®, the Polycom “Triangles” logo and the names and marks associated with Polycom’s products are
trademarks and/or service marks of Polycom, Inc. and are registered and/or common law marks in the United States
and various other countries. All other trademarks are property of their respective owners. No portion hereof may be
reproduced or transmitted in any form or by any means, for any purpose other than the recipient’s personal use, without
the express written permission of Polycom.
Patent Information
The accompanying product is protected by one or more U.S. and foreign patents and/or pending patent applications
held by Polycom, Inc.
Disclaimer
Some countries, states, or provinces do not allow the exclusion or limitation of implied warranties or the limitation of
incidental or consequential damages for certain products supplied to consumers, or the limitation of liability for personal
injury, so the above limitations and exclusions may be limited in their application to you. When the implied warranties
are not allowed to be excluded in their entirety, they will be limited to the duration of the applicable written warranty. This
warranty gives you specific legal rights which may vary depending on local law.
Copyright Notice
Portions of the software contained in this product are:
Copyright © 1998, 1999, 2000 Thai Open Source Software Center Ltd. and Clark Cooper
Copyright © 1998 by the Massachusetts Institute of Technology
Copyright © 1998-2003 The OpenSSL Project
Copyright © 1995-1998 Eric Young (eay@cryptsoft.com). All rights reserved
Copyright © 1995-2002 Jean-Loup Gailly and Mark Adler
Copyright © 1996-2004, Daniel Stenberg, <daniel@haxx.se>
Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated
documentation files (the “Software”), to deal in the Software without restriction, including without limitation the rights to
use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to
whom the Software is furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all copies or substantial portions of the
Software.
THE SOFTWARE IS PROVIDED “AS IS”, WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE
LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
© 2009 Polycom, Inc. All rights reserved.

Polycom, Inc.
4750 Willow Road
Pleasanton, CA 94588-2708
USA
No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for
any purpose, without the express written permission of Polycom, Inc. Under the law, reproducing includes translating
into another language or format.
As between the parties, Polycom, Inc., retains title to and ownership of all proprietary rights with respect to the software
contained within its products. The software is protected by United States copyright laws and international treaty
provision. Therefore, you must treat the software like any other copyrighted material (e.g., a book or sound recording).
Every effort has been made to ensure that the information in this manual is accurate. Polycom, Inc., is not responsible
for printing or clerical errors. Information in this document is subject to change without notice.
ii
About This Guide
The Administrator’s Guide for the SoundPoint IP/SoundStation IP/VVX

family is for administrators who need to configure, customize, manage, and
troubleshoot SoundPoint IP/SoundStation IP/VVX phone systems. This
guide covers the SoundPoint IP 320, 321, 330, 331, 335, 430, 450, 550, 560, 650,
and 670 desktop phones, the SoundStation IP 6000 and 7000 conference
phones, and the Polycom VVX 1500 business media phone.
The following related documents for SoundPoint IP/SoundStation IP/VVX
family are available:
• Quick Start Guides, which describe how to assemble the phones
• Quick User Guides, which describe the most basic features available on
the phones
• User Guides, which describe the basic and advanced features available on
the phones
• Developer’s Guide, which assists in the development of applications that

run on the SoundPoint IP/SoundStation IP/VVX phone’s Microbrowser
• Technical Bulletins, which describe workarounds to existing issues and

provide expanded descriptions and examples
• Release Notes, which describe the new and changed features and fixed
problems in the latest version of the software
For support or service, please contact your Polycom® reseller or go to Polycom
Technical Support at http://www.polycom.com/voicedocumentation/.
Polycom recommends that you record the phone model numbers, software
(both the BootROM and SIP), and partner platform for future reference.
SoundPoint IP/SoundStation IP/VVX models: __________________________
BootROM version: ________________________________________________
SIP Software version: ______________________________________________
Partner Platform: _________________________________________________
iii
Administrator’s Guide for the SoundPoint IP/SoundStation IP/VVX Family
iv
Contents
About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . iii
1 Introducing the SoundPoint IP / SoundStation IP / VVX

Family . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–1
SoundPoint IP Desktop Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–1
SoundStation IP Conference Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–4
Polycom VVX 1500 Business Media Phone . . . . . . . . . . . . . . . . . . . . . . . . . 1–6
Key Features of Your SoundPoint IP / SoundStation IP / VVX Phones 1–6
2 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–1
Where SoundPoint IP / SoundStation IP / VVX Phones Fit . . . . . . . . . . 2–2
Session Initiation Protocol Application Architecture . . . . . . . . . . . . . . . . . 2–3
BootROM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–3
SIP Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–4
Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–5
Resource Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–8
Available Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–8
New Features in SIP 3.2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–14
3 Setting up Your System . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–1

Setting Up the Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2
DHCP or Manual TCP/IP Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2
Supported Provisioning Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–4
Modifying the Network Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 3–6
Setting Up the Provisioning Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–14
Deploying Phones From the Provisioning Server . . . . . . . . . . . . . . . . . . . 3–17
Upgrading SIP Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–21
Supporting SoundPoint IP, SoundStation IP, and Polycom VVX
Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–22
Supporting SoundPoint IP 300, 301, 500, 501, 600 and 601 and
SoundStation IP 4000 Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–23
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Administrator’s Guide for the SoundPoint IP / SoundStation IP / VVX Family
4 Configuring Your System . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1

Setting Up Basic Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1
Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3
Call Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3
Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3
Called Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4
Calling Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4
Missed Call Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5
Connected Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5
Context Sensitive Volume Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5
Customizable Audio Sound Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6
Message Waiting Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7
Distinctive Incoming Call Treatment . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7
Distinctive Ringing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7
Distinctive Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–8
Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–8
Handset, Headset, and Speakerphone . . . . . . . . . . . . . . . . . . . . . . . . . 4–9
Local Contact Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–10
Local Digit Map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13
Microphone Mute . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–14
Soft Key Activated User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–14
Speed Dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15
Time and Date Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15
Idle Display Animation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16
Ethernet Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17
Graphic Display Backgrounds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17
Automatic Off-Hook Call Placement . . . . . . . . . . . . . . . . . . . . . . . . . . 4–19
Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–19
Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20
Local / Centralized Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–21
Call Forward . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22
Directed Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24
Group Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24
Call Park/Retrieve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24
Last Call Return . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25
Setting Up Advanced Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25
Configurable Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26
Multiple Line Keys per Registration . . . . . . . . . . . . . . . . . . . . . . . . . . 4–27
Multiple Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28
Customizable Fonts and Indicators . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28
vi
Contents
Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29

Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29
Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30
Synthesized Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30
Browser and Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31
Real-Time Transport Protocol Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32
Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–33
Corporate Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34
Recording and Playback of Audio Calls . . . . . . . . . . . . . . . . . . . . . . . 4–36
Digital Picture Frame . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–37
Enhanced Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–38
Configurable Soft Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–48
LCD Power Saving . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–52
Shared Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–52
Bridged Line Appearance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–54
Busy Lamp Field . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–55
Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56
Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–57
SIP-B Automatic Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–58
Feature Synchronized Automatic Call Distribution . . . . . . . . . . . . . 4–59
Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–60
Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–64
Microsoft Live Communications Server 2005 Integration . . . . . . . . 4–64
Access URL in SIP Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–68
Static DNS Cache . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–72
Display of Warnings from SIP Headers . . . . . . . . . . . . . . . . . . . . . . . 4–75
Quick Setup of SoundPoint IP / SoundStation IP / VVX Phones . 4–76
Setting Up Audio Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77
Low-Delay Audio Packet Transmission . . . . . . . . . . . . . . . . . . . . . . . 4–77
Jitter Buffer and Packet Error Concealment . . . . . . . . . . . . . . . . . . . . 4–78
Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78
DTMF Tone Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–79
DTMF Event RTP Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–79
Acoustic Echo Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–79
Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80
Background Noise Suppression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–81
Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82
Automatic Gain Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82
IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82
IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82
vii
Voice Quality Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–83

Dynamic Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–84
Treble/Bass Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–84
Setting Up Video Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–85
Video Transmission . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–85
Video Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–86
H.323 Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–87
Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–92
Local User and Administrator Privilege Levels . . . . . . . . . . . . . . . . . 4–92
Custom Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–93
Incoming Signaling Validation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–93
Secure Real-Time Transport Protocol . . . . . . . . . . . . . . . . . . . . . . . . . 4–93
Configuration File Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–95
Digital Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–95
Mutual TLS Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–97
Configuring SoundPoint IP / SoundStation IP / VVX Phones Locally 4–98
5 Troubleshooting Your SoundPoint IP / SoundStation IP / VVX

Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–1
Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2
BootROM Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2
SIP Application Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–3
Status Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4
Log Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–5
Reading a Boot Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–8
Reading an Application Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–9
Reading a Syslog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10
Testing Phone Hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10
Power and Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–11
Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–12
Access to Screens and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–13
Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–14
Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–15
Audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–16
Licensable Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–16
Upgrading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–17
viii
Contents
A Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .A–1

Master Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–2
Application Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–5
Protocol <voIpProt/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–7
Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–23
Localization <lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–27
User Preferences <up/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–31
Tones <tones/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–34
Sampled Audio for Sound Effects <saf/> . . . . . . . . . . . . . . . . . . . . . A–37
Sound Effects <se/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–38
Voice Settings <voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–44
Video Settings <video/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–64
Quality of Service <QOS/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–71
Basic TCP/IP <TCP_IP/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–75
Web Server <httpd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–79
Call Handling Configuration <call/> . . . . . . . . . . . . . . . . . . . . . . . . A–80
Directory <dir/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–85
Presence <pres/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–91
Fonts <font/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92
Keys <key/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–94
Backgrounds <bg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–96
Bitmaps <bitmap/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–100
Indicators <ind/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–100
Event Logging <log/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–104
Security <sec/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–108
License <license/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–113
Provisioning <prov/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–114
RAM Disk <ramdisk/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–115
Request <request/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–115
Feature <feature/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–116
Resource <res/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–118
Microbrowser <mb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–119
Applications <apps/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–122
Peer Networking <pnet/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–125
DNS Cache <dns/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–126
Soft Keys <softkey/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–128
LCD Power Saving <powerSaving/> . . . . . . . . . . . . . . . . . . . . . . . A–132
Per-Phone Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–133
Registration <reg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–134
Calls <call/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–139
ix
Diversion <divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–142

Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–145
Messaging <msg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–149
Network Address Translation <nat/> . . . . . . . . . . . . . . . . . . . . . . A–150
Attendant <attendant/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–151
Roaming Buddies <roaming_buddies/> . . . . . . . . . . . . . . . . . . . . A–154
Roaming Privacy <roaming_privacy/> . . . . . . . . . . . . . . . . . . . . . A–154
User Preferences <up/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–155
Automatic Call Distribution <acd/> . . . . . . . . . . . . . . . . . . . . . . . . A–156
Flash Parameter Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–157
B Session Initiation Protocol (SIP) . . . . . . . . . . . . . . . . . . . . . B–1

RFC and Internet Draft Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–2
Request Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–3
Header Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–4
Response Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–6
Hold Implementation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
Reliability of Provisional Responses . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
Third Party Call Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
SIP for Instant Messaging and Presence Leveraging Extensions . . B–10
Shared Call Appearance Signaling . . . . . . . . . . . . . . . . . . . . . . . . . . . B–10
Bridged Line Appearance Signaling . . . . . . . . . . . . . . . . . . . . . . . . . . B–10
C Miscellaneous Administrative Tasks . . . . . . . . . . . . . . . . . . C–1

Trusted Certificate Authority List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–1
Encrypting Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–4
Changing the Key on the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–6
Adding a Background Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–6
BootROM/SIP Application Dependencies . . . . . . . . . . . . . . . . . . . . . . . . C–9
Migration Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–10
Supported SoundStation IP 7000 / Polycom HDX Software
Interoperability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–10
Multiple Key Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–11
Default Feature Key Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–13
Internal Key Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–19
Assigning a VLAN ID Using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–23
Parsing Vendor ID Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–24
Product, Model, and Part Number Mapping . . . . . . . . . . . . . . . . . . . . . C–26
Disabling PC Ethernet Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–27
x
Contents
Modifying Phone’s Configuration Using the Web Interface . . . . . . . . . C–27

Capturing Phone’s Current Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–30
LLDP and Supported TLVs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–30
Supported TLVs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–32
D Third Party Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–1
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .Index–1
xi
xii
1
Introducing the SoundPoint IP /
SoundStation IP / VVX Family
This chapter introduces the SoundPoint IP / SoundStation IP / VVX family,

which is supported by the software described in this guide.
The SoundPoint IP / SoundStation IP / VVX family provides a powerful, yet
flexible IP communications solution for Ethernet TCP/IP networks, delivering
excellent voice quality. The high-resolution graphic display supplies content
for call information, multiple languages, directory access, and system status.
The SoundPoint IP / SoundStation IP / VVX family supports advanced
functionality, including multiple call and flexible line appearances, HTTPS
secure provisioning, presence, custom ring tones, and local conferencing.
The SoundPoint IP / SoundStation IP / VVX phones are end points in the
overall network topology designed to interoperate with other compatible
equipment including application servers, media servers, internet-working
gateways, voice bridges, and other end points
The following models are described:
• SoundPoint IP Desktop Phones
• SoundStation IP Conference Phones
• Polycom VVX 1500 Business Media Phone

For a list of key features available on the SoundPoint IP / SoundStation IP /
VVX phones running the latest software, refer to Key Features of Your
SoundPoint IP / SoundStation IP / VVX Phones on page 1-6.
SoundPoint IP Desktop Phones

This section describes the current SoundPoint IP desktop phones. For
individual guides, refer to the product literature available at
http://www.polycom.com/voicedocumentation/. Additional options are
also available. For more information, contact your Polycom distributor.
1-1
Note Documentation for the SoundPoint IP 300, 301, 500, 501, 600, and 601 desktop
phones and the SoundStation IP 4000 conference phone is available at
http://www.polycom.com/voicedocumentation/ .
The currently supported desktop phones are:
• SoundPoint IP 320/321/330/331/335
• SoundPoint IP 430
1-2
Introducing the SoundPoint IP / SoundStation IP / VVX Family
• SoundPoint IP 550/560
1-3
SoundStation IP Conference Phones

This section describes the current SoundPoint IP conference phones. For
individual guides, refer to the product literature available at
1-4
The currently supported conference phones are:
• SoundStation IP 6000
• SoundStation IP 7000
1-5
Polycom VVX 1500 Business Media Phone

This section describes the current Polycom VVX 1500 business media phone.
For the individual guide, refer to the product literature available at
Key Features of Your SoundPoint IP / SoundStation IP / VVX

Phones
The key features of the SoundPoint IP / SoundStation IP / VVX phones are:
• Award winning sound quality and full-duplex speakerphone or

conference phone
— Permits natural, high-quality, two-way conversations
— Uses Polycom’s industry leading Acoustic Clarity Technology
• Easy-to-use
— An easy transition from traditional PBX systems into the world of IP
— Up to 18 dedicated hard keys for access to commonly used features
— Up to four context-sensitive soft keys for further menu-driven
activities
1-6
• Platform independent
— Supports multiple protocols and platforms enabling standardization
on one phone for multiple locations, systems and vendors
— Polycom’s support of the leading protocols and industry partners
makes it a future-proof choice
• Field upgradeable
— Upgrade SoundPoint IP / SoundStation IP / VVX as standards
develop and protocols evolve
— Extends the life of the phone to protect your investment
— Application flexibility for call management and new telephony
applications
• Large LCD
— Easy-to-use, easily readable and intuitive interface
— Support of rich application content, including multiple call
appearances, presence and instant messaging, and XML services
— 102 x 23 pixel graphical LCD for the SoundPoint IP
320/321/330/331/335
— 256 x 116 pixel graphical grayscale LCD for the SoundPoint IP 450
— 320 x 160 pixel graphical grayscale LCD for the SoundPoint IP
550/560/650 (supports Asian characters)
— 320 x 160 pixel graphical color LCD for the SoundPoint IP 670
(supports Asian characters)
— 248 x 68 pixel graphical LCD for the SoundStation IP 6000
— 256 x 128 pixel graphical grayscale LCD for the SoundStation IP 7000
— 800 x 480 pixel graphical color LCD for the Polycom VVX 1500 (touch
screen)
• Dual auto-sensing 10/100/1000baseT Ethernet ports

— Leverages existing infrastructure investment
— No re-wiring with existing CAT 5 cabling
— Simplifies installation
— 1000baseT is supported by the SoundPoint IP 560 and 670 and
Polycom VVX 1500 only
• Power over Ethernet (PoE) port or Power pack option

— Built-in IEEE 802.3af PoE port on the SoundPoint IP
320/321/330/331/335, 450, 550, 560, 650, and 670, the SoundStation IP
6000 and 7000, and Polycom VVX 1500 (auto-sensing)
1-7
— Unused pairs on Ethernet port are used to deliver power to the phone
via a wall adapter allowing fewer wires to desktop (for the
SoundStation IP 6000 and 7000 conference phones)
• Multiple language support on most phones

— Set on-screen language to your preference. Select from
Chinese (Simplified), Danish, Dutch, English (Canada, United
Kingdom, and United States), French, German, Italian, Japanese,
Korean, Norwegian, Polish, Portuguese (Brazilian), Russian,
Slovenian, Spanish (International), and Swedish.
— Chinese (Simplified), Japanese, and Korean are not supported on the
SoundPoint IP 320/321/330/331/335 phones.
• Microbrowser
— Supports a subset of XHTML constructs; otherwise runs like any other
Web browser.
• Polycom Browser on the Polycom VVX 1500

— Supports XHTML 1.1 constructs, HTML 4.01, JavaScript, CCS 2.1, and
SVG 1.1 (partial support).
• XML status/control API

— Ability to poll phones for call status and device information.
— Ability to receive telephony notification events.
1-8
2
Overview
This chapter provides an overview of the Session Initiation Protocol (SIP)

application and how the phones fit into the network configuration.
SIP is the Internet Engineering Task Force (IETF) standard for multimedia
communications over IP. It is an ASCII-based, application-layer control
protocol (defined in RFC 3261) that can be used to establish, maintain, and
terminate calls between two or more endpoints. Like other voice over IP
(VoIP) protocols, SIP is designed to address the functions of signaling and
session management within a packet telephony network. Signaling allows call
information to be carried across network boundaries. Session management
provides the ability to control the attributes of an end-to-end call.
For the SoundPoint IP / SoundStation IP / VVX phones to successfully
operate as a SIP endpoint in your network, it must meet the following
requirements:
• A working IP network is established.
• Routers are configured for VoIP.
• VoIP gateways are configured for SIP.
• The latest (or compatible) SoundPoint IP / SoundStation IP / VVX phone

SIP application image is available.
• A call server is active and configured to receive and send SIP messages.
For more information on IP PBX and softswitch vendors, go to
http://www.polycom.com/techpartners1/ .
This chapter contains information on:
• Where SoundPoint IP / SoundStation IP / VVX Phones Fit
• Session Initiation Protocol Application Architecture
• Available Features
• New Features in SIP 3.2

To install your SoundPoint IP / SoundStation IP / VVX phones on the
network, refer to Setting up Your System on page 3-1. To configure your
SoundPoint IP / SoundStation IP / VVX phones with the desired features,
2-1
refer to Configuring Your System on page 4-1. To troubleshoot any problems

with your SoundPoint IP / SoundStation IP / VVX phones on the network,
refer to Troubleshooting Your SoundPoint IP / SoundStation IP / VVX Phones
on page 5-1.
Where SoundPoint IP / SoundStation IP / VVX Phones Fit

The phones connect physically to a standard office twisted-pair (IEEE 802.3)
10/100 megabytes per second Ethernet LAN and send and receive all data
using the same packet-based technology. Since the phone is a data terminal,
digitized audio being just another type of data from its perspective, the phone
is capable of vastly more than traditional business phones. As SoundPoint IP
/ SoundStation IP / VVX phones run the same protocols as your office
personal computer, many innovative applications can be developed without
resorting to specialized technology.
Remote
Boot Server
Internet
PSTN
Remote
Application
Server
Router/
Firewall
PSTN Gateway
10/100 Ethernet
Ethernet Switches
Switch
Voice Bridge
PC PC
Local Application
Server
Or
Local
10/100 Boot Server
Ethernet
Hub
PC
2-2
Overview
Session Initiation Protocol Application Architecture

The software architecture of the SIP application is made of 4 basic components:
• BootROM—loads first when the phone is powered on
• SIP Application—software that makes the device a phone
• Configuration—configuration parameters stored in separate files
• Resource Files—optional, needed by some of the advanced features
BootROM
The BootROM is a small application that resides in the flash memory on the
phone. All phones come from the factory with a BootROM pre-loaded.
The BootROM performs the following tasks in order:
1. Performs a power on self test (POST).
2. (Optional) Allows you to enter the setup menu where various network
and provisioning options can be set.
The BootROM software controls the user interface when the setup menu
is accessed.
3. Requests IP settings and accesses the provisioning server (or boot server)
to look for any updates to the BootROM application.
If updates are found, they are downloaded and saved to flash memory,
eventually overwriting itself after verifying the integrity of the download.
4. If a new BootROM is downloaded, formats the file system clearing out
any application software and configuration files that may have been
present.
2-3
5. Downloads the master configuration file.

This file is either called <MAC-address>.cfg or 000000000000.cfg . This file
is used by the BootROM and the application for a list of other files that are
needed for the operation of the phone.
6. Examines the master configuration file for the name of the application
file, and then looks for this file on the provisioning server.
If the copy on the provisioning server is different than the one stored in
flash memory or there is no file stored in flash memory, the application file
is downloaded.
7. Extracts the application from flash memory.
8. Installs the application into RAM, then uploads a log file with events
from the boot cycle.
The BootROM will then terminate, and the application takes over.
SIP Application
The SIP application manages the VoIP stack, the digital signal processor (DSP),
the user interface, and the network interaction. The SIP application manages
everything to do with the phone’s operation.
The application is a single file binary image and contains a digital signature to
prevent tampering or loading rogue software images.
There is a new image file in each release of software.
The application performs the following tasks in order:
1. Downloads system, per-phone configuration, and resource files.
These files are called sip.cfg and phone1.cfg by default. You can
customize the filenames.
2. Controls all aspects of the phone.
3. Uploads log files.
BootROM and SIP Application Wrapper

Both the BootROM and the application run on multiple platforms (meaning all
previously released versions of hardware that are still supported).
Current build archives have both split and combined images, so it is up to the
administrator which model(s) to support. Using split files saves a lot of
internal network traffic during reboots and updates.
2-4
Overview
Configuration
The SoundPoint IP / SoundStation IP / VVX phones can be configured
automatically through files stored on a central provisioning server, manually
through the phone’s local UI or web interface, or by using a combination of the
automatic and manual methods.
The recommended method for configuring phones is automatically through a
central provisioning server, but if one is not available, the manual method will
allow changes to most of the key settings.
Warning Configuration files should only be modified by a knowledgeable system

administrator. Applying incorrect parameters may render the phone unusable. The
configuration files which accompany a specific release of the SIP software must be
used together with that software. Failure to do this may render the phone unusable.
Note You can make changes to the configuration files through the web interface to the
phone. Using your chosen browser, enter the phone’s IP address as the browser
address. For more information, refer to Modifying Phone’s Configuration Using the
Web Interface on page C-27.
Changes made through the web interface are written to the override file (highest
priority). These changes remain active and will take precedence over the
configuration files stored on the provisioning server until Reset Local Config is
performed.
The phone configuration files consist of:
• Master Configuration Files
• Application Configuration Files
• Override Files
This section also contains information on:
• Central Provisioning
• Manual Configuration
Master Configuration Files

The master configuration files can be one of:
• Specified master configuration file
• Per-phone master configuration file
• Default master configuration file

For more information, refer to Master Configuration Files on page A-2.
2-5
Application Configuration Files

Typically, the files are arranged in the following manner although parameters
may be moved around within the files and the filenames themselves can be
changed as needed. These files dictate the behavior of the phone once it is
running the executable specified in the master configuration file.
The application files are:
• Application—It contains parameters that affect the basic operation of the

phone such as voice codecs, gains, and tones and the IP address of an
application server. All phones in an installation usually share this category
of files. Polycom recommends that you create another file with your
organization’s modifications. If you must change any Polycom templates,
back them up first. By default, sip.cfg is included.
• Per-phone—It contains parameters unique to a particular phone user.

Typical parameters include:
— display name
— unique addresses
Each phone in an installation usually has its own customized version of
user files derived from Polycom templates. By default, phone1.cfg is
included.
Override Files
This file contains all changes that are made by a user through the their phone
(for example, time/date formats, ring types, and backlight intensity). The file
allows the phone to keep user preferences through reboots and upgrades.
There is an option to clear the override file available to the system
administrator—press the Menu key, and then select Settings > Advanced >
Admin Settings > Reset to Default > Reset Local Config. You will be
prompted to enter the administrative password.
Central Provisioning
The phones can be centrally provisioned from a provisioning server through a
system of global and per-phone configuration files. The provisioning server
also facilitates automated application upgrades, logging, and a measure of
fault tolerance. Multiple redundant provisioning servers can be configured to
improve reliability.
In the central provisioning method, there are two major classifications of
configuration files:
• System configuration files
• Per-phone configuration files
2-6
Overview
Parameters can be stored in the files in any order and can be placed in any
number of files. The default is to have 2 files, one for per-phone setting and one
for system settings. The per-phone file is typically loaded first, and could
contain system level parameters, letting you override that parameter for a
given user. For example, it might be desirable to set the default CODEC for a
remote user differently than for all the users who reside in the head office. By
adding the CODEC settings to a particular user’s per-phone file, the values in
the system file are ignored.
Note Verify the order of the configuration files. Parameters in the configuration file loaded
first will overwrite those in later configuration files.
The following figure shows one possible layout of the central provisioning
method.
Boot Server
config overrides event log

contact directory files
master config file

application binary
config files
dictionary files
user interface
resource files
license files
SoundPoint IP SIP
Local User Interface Local
MAC 00:04:f2:00:29:99 Web Server
Manual Configuration
When the manual configuration method is employed, any changes made are
stored in a configuration override file. This file is stored on the phone, but a
copy will also be uploaded to the central provisioning server if one is being
used. When the phone boots, this file is loaded by the application after any
centrally provisioned files have been read, and its settings will override those
in the centrally provisioned files.
2-7
This can create a lot of confusion about where parameters are being set, and so
it is best to avoid using the manual method unless you have good reason to do
so.
Resource Files
In addition to the application and the configuration files, the phones may
require resource files that are used by some of the advanced features. These
files are optional, but if the particular feature is being employed, these files are
required.
Some examples of resource files include:
• Language dictionaries
• Custom fonts
• Ring tones
• Synthesized tones
• Contact directories
Note If you need to remove the resource files from a phone at some later date—for
example, you are giving the phone to a new user—instructions on how to put the
phone into the factory default state can be found in “Quick Tip 18298: Resetting and
Rebooting SoundPoint IP, SoundStation IP, and Polycom VVX 1500 Phones“ at
http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pu
b.html .
Available Features
This section provides information about the features available on the
SoundPoint IP / SoundStation IP / VVX phones:
• Basic User Features

— Automatic Off-Hook Call Placement—Supports an optional
automatic off-hook call placement feature for each registration.
— Call Forward—Provides a flexible call forwarding feature to forward
calls to another destination.
— Call Hold—Pauses activity on one call so that the user may use the
phone for another task, such as making or receiving another call.
— Call Log—Contains call information such as remote party
identification, time and date, and call duration in three separate lists,
missed calls, received calls, and placed calls on most platforms.
— Call Park/Retrieve—An active call can be parked. A parked call can
be retrieved by any phone.
2-8
Overview
— Call Timer—A separate call timer, in hours, minutes, and seconds, is

maintained for each distinct call in progress.
— Call Transfer—Call transfer allows the user to transfer a call in
progress to some other destination.
— Call Waiting—When an incoming call arrives while the user is active
on another call, the incoming call is presented to the user visually on
the display and a configurable sound effect will be mixed with the
active call audio.
— Called Party Identification—The phone displays and logs the identity
of the party specified for outgoing calls.
— Calling Party Identification—The phone displays the caller identity,
derived from the network signaling, when an incoming call is
presented, if information is provided by the call server.
— Connected Party Identification—The identity of the party to which the
user has connected is displayed and logged, if the name is provided
by the call server.
— Context Sensitive Volume Control—The volume of user interface
sound effects, such as the ringer, and the receive volume of call audio
is adjustable.
— Customizable Audio Sound Effects—Audio sound effects used for
incoming call alerting and other indications are customizable.
— Directed Call Pick-Up and Group Call Pick-Up—Calls to another
phone can be picked up by dialing the extension of the other phone.
Calls to another phone within a pre-defined group can be picked up
without dialing the extension of the other phone.
— Distinctive Call Waiting—Calls can be mapped to distinct call waiting
types.
— Distinctive Incoming Call Treatment—The phone can automatically
apply distinctive treatment to calls containing specific attributes.
— Distinctive Ringing—The user can select the ring type for each line
and the ring type for specific callers can be assigned in the contact
directory.
— Do Not Disturb—A do-not-disturb feature is available to temporarily
stop all incoming call alerting.
— Graphic Display Backgrounds—A picture or design displayed on the
background of the graphic display.
— Handset, Headset, and Speakerphone—SoundPoint IP phones come
standard with a handset and a dedicated headset connection (headset
not supplied). All SoundPoint IP, SoundStation IP, and Polycom VVX
phones have full-duplex speakerphones.
— Idle Display Animation—All phones can display a customized
animation on the idle display in addition to the time and date.
2-9
— Last Call Return—The phone allows call server-based last call return.
— Local / Centralized Conferencing—The phone can conference
together the local user with the remote parties of two independent
calls and can support centralized conferences for which external
resources are used such as a conference bridge. The advanced aspects
of conferencing are part of the Productivity Suite.
— Local Contact Directory—The phone maintains a local contact
directory that can be downloaded from the provisioning server and
edited locally. Any edits to the Contact Directory made on the phone
are saved to the provisioning server as a backup.
— Local Digit Map—The phone has a local digit map to automate the
setup phase of number-only calls.
— Message Waiting Indication—The phone will flash a message-waiting
indicator (MWI) LED when instant messages and voice messages are
waiting.
— Microphone Mute—When the microphone mute feature is activated,
visual feedback is provided.
— Missed Call Notification—The phone can display the number of calls
missed since the user last looked at the Missed Calls list.
— Soft Key Activated User Interface—The user interface makes
extensive use of intuitive, context-sensitive soft key menus.
— Speed Dial—The speed dial system allows calls to be placed quickly
from dedicated keys as well as from a speed dial menu.
— Time and Date Display—Time and date can be displayed in certain
operating modes such as when the phone is idle and during a call.
• Advanced Features
— Access URL in SIP Message—Ability for the SoundPoint IP phones to
be able to receive a URL inside a SIP message (for example, as a SIP
header extension in a SIP INVITE) and subsequently access that given
URL in the Microbrowser.
— SIP-B Automatic Call Distribution—Supports ACD agent available
and unavailable and allows ACD login and logout. Requires call
server support.
— Bridged Line Appearance—Calls and lines on multiple phones can be
logically related to each other. Requires call server support.
— Busy Lamp Field—Allows monitoring the hook status and remote
party information of users through the busy lamp field (BLF) LEDs
and displays on an attendant console phone. This feature may require
call server support.
— Configurable Feature Keys—Certain key functions can be changed
from the factory defaults.
2 - 10
Overview
— Configurable Soft Keys—Allows users to create their own soft keys

and have them displayed with or without the standard SoundPoint IP
and SoundStation IP soft keys.
— Corporate Directory—The phone can be configured to access your
corporate directory if it has a standard LDAP interface. This feature is
part of the Productivity Suite.
— Customizable Fonts and Indicators—The phone’s user interface can
be customized by changing the fonts and graphic icons used on the
display and the LED indicator patterns.
— Display of Warnings from SIP Headers—Displays a “pop-up” to user
that is found in the Warning Field from a SIP header.
— Downloadable Fonts—New fonts can be loaded onto the phone.
— Enhanced Busy Lamp Field—Allows an attendant to see a remote line
that is Ringing and answer a remote ringing call using a single
key-press. Also allows the attendant to view the caller-id of remote
active and ringing calls. This feature may require call server support.
— Enhanced Feature Keys—Allows users to redefine soft keys to suit
their needs. In SIP 3.0, this feature required a license key.
— Instant Messaging—Supports sending and receiving instant text
messages.
— Browser and Microbrowser—The SoundPoint IP 430, 450, 550, 560,
600, 601, 650, and 670 desktop phones, the SoundStation IP 6000, and
7000 conference phones, and the Polycom VVX 1500 phones (pre-SIP
3.2.2) support an XHTML microbrowser.
— Microsoft Live Communications Server 2005
Integration—SoundPoint IP and SoundStation IP phones can be used
with Microsoft Live Communications Server 2005 and Microsoft
Office Communicator to help improve business efficiency and
increase productivity and to share ideas and information immediately
with business contacts. Requires call server support.
— Multilingual User Interface—All phones have multilingual user
interfaces.
— Multiple Call Appearances—The phone supports multiple concurrent
calls. The hold feature can be used to pause activity on one call and
switch to another call.
— Multiple Line Keys per Registration—More than one line key can be
allocated to a single registration.
— Multiple Registrations—SoundPoint IP desktop phones and Polycom
VVX 1500 phones support multiple registrations per phone. However,
SoundStation IP conference phones support a single registration.
— Network Address Translation—The phones can work with certain
types of network address translation (NAT).
2 - 11
— Presence—Allows the phone to monitor the status of other

users/devices and allows other users to monitor it. Requires call
server support.
— Real-Time Transport Protocol Ports—The phone treats all real- time
transport protocol (RTP) streams as bi-directional from a control
perspective and expects that both RTP end points will negotiate the
respective destination IP addresses and ports.
— Recording and Playback of Audio Calls — Recording and playback
allows the user to record any active conversation using the phone on
a USB device. The files are date and time stamped for easy archiving
and can be played back on the phone or on any computer with a media
playback program that supports the .wav format. This feature is part
of the Productivity Suite.
— Server Redundancy—Server redundancy is often required in VoIP
deployments to ensure continuity of phone service for events where
the call server needs to be taken offline for maintenance, the server
fails, or the connection from the phone to the server fails.
— Shared Call Appearances—Calls and lines on multiple phones can be
logically related to each other. Requires call server support.
— Static DNS Cache—Set up a static DNS cache and provide for negative
caching.
— Synthesized Call Progress Tones—In order to emulate the familiar
and efficient audible call progress feedback generated by the PSTN
and traditional PBX equipment, call progress tones are synthesized
during the life cycle of a call. Customizable for certain regions, for
example, Europe has different tones from North America.
— Voice Mail Integration—Compatible with voice mail servers.
• Audio Features
— Acoustic Echo Cancellation—Employs advanced acoustic echo
cancellation for hands-free operation.
— Audio Codecs—Supports a wide range of industry standard audio
codecs.
— Automatic Gain Control—Designed for hands-free operation, boosts
the transmit gain of the local user in certain circumstances.
— Background Noise Suppression—Designed primarily for hands-free
operation, reduces background noise to enhance communication in
noisy environments.
— Comfort Noise Fill—Designed to help provide a consistent noise level
to the remote user of a hands-free call.
— DTMF Event RTP Payload—Conforms to RFC 2833, which describes
a standard RTP-compatible technique for conveying DTMF dialing
and other telephony events over an RTP media stream.
2 - 12
Overview
— DTMF Tone Generation—Generates dual tone multi-frequency

(DTMF) tones in response to user dialing on the dial pad.
— Dynamic Noise Reduction— Provides maximum microphone
sensitivity, while automatically reducing background noise on
SoundStation IP 7000 conference phones.
— IEEE 802.1p/Q—The phone will tag all Ethernet packets it transmits
with an 802.1Q VLAN header.
— IP Type-of-Service—Allows for the setting of TOS settings.
— Jitter Buffer and Packet Error Concealment—Employs a
high-performance jitter buffer and packet error concealment system
designed to mitigate packet inter-arrival jitter and out-of-order or lost
(lost or excessively delayed by the network) packets.
— Low-Delay Audio Packet Transmission—Designed to minimize
latency for audio packet transmission.
— Treble/Bass Controls—Equalizes the tone of the high and low
frequency sound from the speakers on SoundStation IP 7000
conference phones.
— Voice Activity Detection—Conserves network bandwidth by
detecting periods of relative “silence” in the transmit data path and
replacing that silence efficiently with special packets that indicate
silence is occurring.
— Voice Quality Monitoring—Generates various quality metrics
including MOS and R-factor for listening and conversational quality.
This feature is part of the Productivity Suite.
• Security Features
— Local User and Administrator Privilege Levels—Several local settings
menus are protected with two privilege levels, user and
administrator, each with its own password.
— Configuration File Encryption—Confidential information stored in
configuration files must be protected (encrypted). The phone can
recognize encrypted files, which it downloads from the provisioning
server and it can encrypt files before uploading them to the
provisioning server.
— Custom Certificates—When trying to establish a connection to a
provisioning server for application provisioning, the phone trusts
certificates issued by widely recognized certificate authorities (CAs).
— Incoming Signaling Validation—Levels of security are provided for
validating incoming network signaling.
— Secure Real-Time Transport Protocol—Encrypting audio streams to
avoid interception and eavesdropping.
2 - 13
For more information on each feature and its associated configuration

parameters, see the appropriate section in Configuring Your System on page
4-1.
New Features in SIP 3.2

Note The SoundPoint IP 300 and 500 phones will be supported on the latest
maintenance patch release of the SIP 2.1 software stream—currently SIP 2.1.4 .
Any new features introduced after SIP 2.1.4 are not supported. Refer to the SIP 2.1
Administrator Guide, which is available at
http://www.polycom.com/global/documents/support/setup_maintenance/products/v
oice/sip_2.1_addendum_to_sip_2.0_administrator%27s_guide.pdf .
The SoundPoint IP 301, 501, 600, and 601 and the SoundStation IP 4000 phones
will be supported on the latest maintenance patch release of the SIP 3.1 software
stream—currently SIP 3.1.3 . Any new features introduced after 3.1.3 are not
supported. Configuration parameters related to these phones will be removed from
the sip.cfg and phone1.cfg files in the next major release. To administer these
phones, refer to the SIP 3.1 Administrator’s Guide, which is available at
http://www.polycom.com/voicedocumentation/ .
The following new features were introduced in SIP 3.1.2:
• Feature Synchronized Automatic Call Distribution—Supports ACD agent

available and unavailable and allows ACD sign in and sign out. Requires
call server support.
• Quick Setup of SoundPoint IP / SoundStation IP / VVX

Phones—Simplifies the process of entering provisioning server
parameters.
The following new feature enhancement was introduced in SIP 3.1.3:
• Corporate Directory—The phone’s user interface to access your corporate

directory has changed. Also Microsoft ADAM and SunLDAP are also
supported in addition to Active Directory and OpenLDAP.
The following new features were introduced in SIP 3.2:
• LLDP and Supported TLVs—Support for Link Layer Discovery Protocol

(LLDP) and media extensions (LLDP-MED) such as VLAN configuration.
For provisioning information, refer to Ethernet Menu on page 3-12.
• iLBC added to Audio Codecs—Support for Internet Low Bitrate Codec

(iLBC) added for the SoundPoint IP 32x/33x, 450, 550, 560, 650, and 670, d
SoundStation IP 6000 and 700, and Polycom VVX 1500.
• Video Codecs—Support the standard video codecs on the Polycom VVX

1500 phones.
2 - 14
Overview
• Mutual TLS Authentication—Support for phone authentication of the

server and server authentication of the phone.
• Digital Certificates— Support for digital certificates and associated private

keys on certain models of SoundPoint IP, SoundStation IP, and Polycom
VVX phones.
• Capturing Phone’s Current Screen—Allows the phone’s current display to

be displayed in a web browser.
The following existing features were changed in SIP 3.2:
• Busy Lamp Field— The BLF feature has been enhanced as follows:
— To provide individual subscription-based BLF monitoring (without
requiring a centralized resource list to be maintained by the call
server.
— To allow the single button ‘remote pick-up’ feature to be implemented
using Directed Call Pick-Up using SIP signaling as well as the star
code method supported in SIP 3.1 .
• Secure Real-Time Transport Protocol—Information has been transferred

from the “Technical Bulletin 25751: Secure Real-Time Transport Protocol
on SoundPoint IP Phones” to this guide.
The following new features were introduced in SIP 3.2.2:
• H.323 Protocol—Support for the H.323 protocol for the Polycom VVX 1500
phone.
• Browser and Microbrowser—Support for a Webkit-style browser for the

Polycom VVX 1500 phone only.
Documentation of the newly released SoundPoint IP 321/331/335 and 450
desktop phones and Polycom VVX 1500 business media phone has also been
added.
Note When SoundPoint IP 32x/33x is used in this guide, it includes the SoundPoint IP
320, 321, 330, 331, and 335 phones.
2 - 15
2 - 16
3
Setting up Your System
Your SoundPoint IP / SoundStation IP / VVX SIP phone is designed to be

used like a regular phone on a public switched telephone network (PSTN).
This chapter provides basic instructions for setting up your SoundPoint IP /
SoundStation IP / VVX phones. This chapter contains information on:
• Setting Up the Network
• Setting Up the Provisioning Server
• Deploying Phones From the Provisioning Server
• Upgrading SIP Application

Because of the large number of optional installations and configurations that
are available, this chapter focuses on one particular way that the SIP
application and the required external systems might initially be installed and
configured in your network.
For more information on configuring your system, refer to Configuring Your
System on page 4-1. For more information on the configuration files required
for setting up your system, refer to Configuration Files on page A-1.
For installation and maintenance of SoundPoint IP / SoundStation IP / VVX phones,

the use of a provisioning server is strongly recommended. This allows for flexibility
in installing, upgrading, maintaining, and configuring the phone. Configuration, log,
and directory files are normally located on this server. Allowing the phone write
access to the server is encouraged.
The phone is designed such that, if it cannot locate a provisioning server when it
boots up, it will operate with internally saved parameters. This is useful for
occasions when the provisioning server is not available, but is not intended to be
used for long-term operation of the phones.
However, if you want to register a single SoundPoint IP / SoundStation IP / VVX
phone, refer to “Quick Tip 44011: Register Standalone SoundPoint IP, SoundStation
IP, and Polycom VVX 1500 Phones“ at
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_Technical_Bulle
tins_pub.html .
3-1
Setting Up the Network

Regardless of whether or not you will be installing a centrally provisioned
system, you must perform basic TCP/IP network setup, such as IP address
and subnet mask configuration, to get your organization’s phones up and
running.
The SIP application uses the network to query the provisioning server for
upgrades, which is an optional process that will happen automatically when
properly deployed. For more information on the basic network settings, refer
to DHCP or Manual TCP/IP Setup on page 3-2.
The BootROM on the phone performs the provisioning functions of
downloading the BootROM, the <MACaddress>.cfg file, and the SIP
application, and uploading log files. For more information, refer to Supported
Provisioning Protocols on page 3-4.
Basic network settings can be changed during BootROM download using the
BootROM’s setup menu. A similar menu system is present in the application
for changing the same network parameters. For more information, refer to
Modifying the Network Configuration on page 3-6.
DHCP or Manual TCP/IP Setup

Basic network settings can be derived from DHCP, or entered manually using
the phone’s LCD-based user interface, or downloaded from configuration
files.
Polycom recommends using DHCP where possible to eliminate repetitive manual

data entry.
The following table shows the manually entered networking parameters that
may be overridden by parameters obtained from a DHCP server, an alternate
DHCP server, or configuration file:
Alternate Configuration File Local

Parameter DHCP Option DHCP DHCP (application only) FLASH
D priority when more than one source exists D
1 2 3 4
IP address 1 • - - •
subnet mask 1 • - - •
IP gateway 3 • - - •
3-2
Alternate Configuration File Local

Parameter DHCP Option DHCP DHCP (application only) FLASH
Refer to DHCP • • - •
boot server Menu on page
address 3-8
151 • - - •
Note: This value
SIP server address is configurable.
SNTP server 42 then 4 • - • •

address
SNTP GMT offset 2 • - • •
DNS server IP 6 • - - •
address
alternate DNS 6 • - - •
server IP address
DNS domain 15 • - - •
Refer to DHCP Warning: Link Layer Discovery Protocol (LLDP) overrides Cisco
Menu on page Discovery Protocol (CDP). CDP overrides Local FLASH which
VLAN ID 3-8 overrides DHCP VLAN Discovery.
For more information on DHCP options, go to

http://www.ietf.org/rfc/rfc2131.txt?number=2131 or
http://www.ietf.org/rfc/rfc2132.txt?number=2132.
Note The configuration file value for SNTP server address and SNTP GMT offset can
be configured to override the DHCP value. Refer to
tcpIpApp.sntp.address.overrideDHCP in Time Synchronization <sntp/> on page
A-75.
The CDP Compatibility value can be obtained from a connected Ethernet switch if
the switch supports CDP.
In the case where you do not have control of your DHCP server or do not have
the ability to set the DHCP options, an alternate method of automatically
discovering the provisioning server address is required. Connecting to a
secondary DHCP server that responds to DHCP INFORM queries with a
requested provisioning server value is one possibility. For more information,
refer to http://www.ietf.org/rfc/rfc3361.txt?number=3361 and
http://www.ietf.org/rfc/rfc3925.txt?number=3925.
3-3
Supported Provisioning Protocols

The BootROM performs the provisioning functions of downloading
configuration files, uploading and downloading the configuration override
file and user directory, and downloading the dictionary and uploading log
files.
The protocol that will be used to transfer files from the provisioning server
depends on several factors including the phone model and whether the
BootROM or SIP application stage of provisioning is in progress. By default,
the phones are shipped with FTP enabled as the provisioning protocol. If an
unsupported protocol is specified, this may result in a defined behavior (see
the table below for details of which protocol the phone will use). The Specified
Protocol listed in the table can be selected in the Server Type field or the Server
Address can include a transfer protocol, for example http://usr:pwd@server
(refer to Server Menu on page 3-10). The boot server address can be an IP
address, domain string name, or URL. The boot server address can also be
obtained through DHCP. Configuration file names in the <MACaddress>.cfg
file can include a transfer protocol, for example
https://usr:pwd@server/dir/file.cfg. If a user name and password are
specified as part of the server address or file name, they will be used only if the
server supports them.
Note A URL should contain forward slashes instead of back slashes and should not
contain spaces. Escape characters are not supported. If a user name and password
are not specified, the Server User and Server Password will be used (refer to
Server Menu on page 3-10).
Protocol used by Protocol used by

BootROM SIP Software
IP 32x, 33x, 430, IP 32x, 33x, 430,

450, 550, 560, 650, 450, 550, 560, 650,
Specified 670, 6000, 7000 670, 6000, 7000
Protocol VVX 1500 VVX 1500
FTP FTP FTP
TFTP TFTP TFTP
HTTP HTTP HTTP
HTTPS HTTP HTTPS
Note There are two types of FTP methods—active and passive. The SIP application is
not compatible with active FTP. Secure provisioning was implemented in a previous
release.
3-4
Note Setting Option 66 to tftp://192.168.9.10 has the effect of forcing a TFTP download.
Using a TFTP URL (for example, tftp://provserver.polycom.com) has the same
effect.
Note Both digest and basic authentication are supported when using HTTP/S for the SIP
application. Only digest authentication is supported when using HTTP by the
BootROM. If the Server Type is configured as HTTPS, the BootROM will contact
the same address and apply the same username and password to authentication
challenges only the protocol used will be HTTP. No SSL negotiation will take place,
so servers that do not allow unsecured HTTP connections will not be able to
provision files.
For downloading the BootROM and application images to the phone, the
secure HTTPS protocol is not available. To guarantee software integrity, the
BootROM will only download cryptographically signed BootROM or
application images. For HTTPS, widely recognized certificate authorities are
trusted by the phone (refer to Trusted Certificate Authority List on page C-1)
and custom certificates can be added to the phone (refer to “Technical Bulletin
17877: Using Custom Certificates With SoundPoint IP, SoundStation IP, and É
Phones“ at
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T
echnical_Bulletins_pub.html .
As of SIP 3.2, Mutual Transport Layer Security (TLS) authentication is
available. For more information, refer to Mutual TLS Authentication on page
4-97.
Note If you want to use digest authentication against the Microsoft Internet Information
Services server:
• Use Microsoft Internet Information Server 6.0 or later.
• Digest authentication needs the user name and password to be saved in
reversible encryption.
• The user account on the server must have administrative privileges.
• The wildcard must be set as MIME type; otherwise the phone will not download
*.cfg, *.ld and other required files. This is due to the fact that the Microsoft
Internet Information Server cannot recognize these extensions and will return a
“File not found” error. To configure wildcard for MIME type, refer to
http://support.microsoft.com/kb/326965 .
For more information, refer to
http://www.microsoft.com/technet/prodtechnol/WindowsServer2003/Library/IIS/809
552a3-3473-48a7-9683-c6df0cdfda21.mspx?mfr=true .
3-5
Modifying the Network Configuration

You can access the network configuration menu:
• During BootROM Phase. The network configuration menu is accessible

during the auto-boot countdown of the BootROM phase of operation.
Press the Setup soft key to launch the main menu.
• During Application Phase. The network configuration menu is accessible

from the phone’s main menu. Select Menu>Settings>Advanced>Admin
Settings>Network Configuration. Advanced Settings are locked by
default. Enter the administrator password to unlock. The factory default
password is 456. Polycom recommends that you change the
administrative password from the default value.
Phone network configuration parameters may be modified by means of:
• Main Menu
• DHCP Menu
• Server Menu
• Ethernet Menu
• Syslog Menu
Use the soft keys, the arrow keys, the Select and Delete keys to make changes.
Certain parameters are read-only due to the value of other parameters. For
example, if the DHCP Client parameter is enabled, the Phone IP Addr and
Subnet Mask parameters are dimmed or not visible since these are guaranteed
to be supplied by the DHCP server (mandatory DHCP parameters) and the
statically assigned IP address and subnet mask will never be used in this
configuration.
Resetting to Factory Defaults

The basic network configuration referred to in the subsequent sections can be
reset to factory defaults using a menu selection from the Advanced Settings
menu or using a multiple key combination described in Multiple Key
Combinations on page C-11.
3-6
Main Menu
The following configuration parameters can be modified on the main setup
menu:
Name Possible Values Description
DHCP Client Enabled, Disabled If enabled, DHCP will be used to obtain the parameters
discussed in DHCP or Manual TCP/IP Setup on page
3-2.
DHCP Menu Refer to DHCP Menu on page 3-8.

Note: Disabled when DHCP client is disabled.
Phone IP Address dotted-decimal IP address Phone’s IP address.

Note: Disabled when DHCP client is enabled.
Subnet Mask dotted-decimal subnet Phone’s subnet mask.

mask Note: Disabled when DHCP client is enabled.
IP Gateway dotted-decimal IP address Phone’s default router.
Server Menu Refer to Server Menu on page 3-10.
SNTP Address dotted-decimal IP address Simple Network Time Protocol (SNTP) server from
OR which the phone will obtain the current time.
domain name string
GMT Offset -13 through +12 Offset of the local time zone from Greenwich Mean
Time (GMT) in half hour increments.
DNS Server dotted-decimal IP address Primary server to which the phone directs Domain
Name System (DNS) queries.
DNS Alternate Server dotted-decimal IP address Secondary server to which the phone directs Domain
Name System queries.
DNS Domain domain name string Phone’s DNS domain.
Ethernet Refer to Ethernet Menu on page 3-12.
EM Power Enabled, Disabled This parameter is relevant if the phone gets Power over
Ethernet (PoE). If enabled, the phone will set power
requirements in CDP to 12W so that up to three
Expansion Modules (EM) can be powered. If disabled,
the phone will set power requirements in CDP to 5W
which means no Expansion Modules can be powered (it
will not work).
Syslog Refer to Syslog Menu on page 3-13.
3-7
Note A parameter value of “???” indicates that the parameter has not yet been set and
saved in the phone’s configuration. Any such parameter should have its value set
before continuing.
The EM Power parameter is only available on SoundPoint IP 650 and 670 phones.
Note To switch the text entry mode on the SoundPoint IP 32x/33x, press the #. You may
want to use URL or IP address modes when entering server addresses.
DHCP Menu
The DHCP menu is accessible only when the DHCP client is enabled. The
following DHCP configuration parameters can be modified on the DHCP
menu:
Possible
Name Values Description
Boot Server 0=Option 66 The phone will look for option number 66 (string type) in the
response received from the DHCP server. The DHCP server
should send address information in option 66 that matches one
of the formats described for Server Address in the next
section, Server Menu.
If the DHCP server sends nothing, the following scenarios are
possible:
• If a boot server value is stored in flash memory and the
value is not “0.0.0.0”, then the value stored in flash is used.
• Otherwise the phone sends out a DHCP INFORM query.
- If a single alternate DHCP server responds, this is

functionally equivalent to the scenario where the primary
DHCP server responds with a valid boot server value.
- If no alternate DHCP server responds, the INFORM query

process will retry and eventually time out.
3-8
Possible
Boot Server (continued) 1=Custom The phone will look for the option number specified by the Boot
Server Option parameter (below), and the type specified by
the Boot Server Option Type parameter (below) in the
response received from the DHCP server.
possible:

DHCP server responds with a valid boot server value.

2=Static The phone will use the boot server configured through the
Server Menu. For more information, refer to the next section,
Server Menu.
3=Custom+Option The phone will first use the custom option if present or use
66 Option 66 if the custom option is not present.
possible:

DHCP server responds with a valid boot server value. The
phone prefers the custom option value over the Option 66
value, but if no custom option is given, the phone will use
the Option 66 value.

Boot Server Option 128 through 254 When the boot server parameter is set to Custom, this
(Cannot be the parameter specifies the DHCP option number in which the
same as VLAN ID phone will look for its boot server.
Option)
Boot Server Option Type 0=IP Address, When the Boot Server parameter is set to Custom, this
1=String parameter specifies the type of the DHCP option in which the
phone will look for its boot server. The IP Address must specify
the boot server. The String must match one of the formats
described for Server Address in the next section, Server
Menu.
3-9
Possible
VLAN Discovery 0=Disabled No VLAN discovery through DHCP.

(default)
1=Fixed Use predefined DHCP vendor-specific option values of 128,

144, 157 and 191. If this is used, the VLAN ID Option field will
be ignored
2=Custom Use the number specified in the VLAN ID Option field as the
DHCP private option value.
VLAN ID Option 128 through 254 The DHCP private option value (when VLAN Discovery is set
(Cannot be the to Custom).
same as Boot For more information, refer to Assigning a VLAN ID Using
Server Option) DHCP on page C-23.
(default is 129)
Note If multiple alternate DHCP servers respond:

• The phone should gather the responses from alternate DHCP servers.
• If configured for Custom+Option66, the phone will select the first response that
contains a valid "custom" option value.
• If none of the responses contain a "custom" option value, the phone will select
the first response that contains a valid “option66” value.
Server Menu
The following server configuration parameters can be modified on the Server
menu:
Server Type 0=FTP, 1=TFTP, 2=HTTP, The protocol that the phone will use to obtain
3=HTTPS, 4=FTPS, 5=Invalid configuration and phone application files from the
provisioning server. Refer to Supported Provisioning
Protocols on page 3-4.
Note: Active FTP is not supported for BootROM version
3.0 or later. Passive FTP is still supported.
Note: Only implicit FTPS is supported.
3 - 10
Server Address dotted-decimal IP address The provisioning server to use if the DHCP client is
OR disabled, the DHCP server does not send a boot server
domain name string option, or the Boot Server parameter is set to Static. The
OR phone can contact multiple IP addresses per DNS name.
URL These redundant provisioning servers must all use the
All addresses can be followed same protocol. If a URL is used it can include a user
by an optional directory and name and password. Refer to Supported Provisioning
optional file name. Protocols on page 3-4. A directory and the master
configuration file can be specified.
Note: ":", "@", or "/" can be used in the user name or
password these characters if they are correctly escaped
using the method specified in RFC 1738.
Server User any string The user name used when the phone logs into the server
(if required) for the selected Server Type.
Note: If the Server Address is a URL with a user name,
this will be ignored.
Server Password any string The password used when the phone logs in to the server
if required for the selected Server Type.
Note: If the Server Address is a URL with user name and
password, this will be ignored.
File Transmit Tries 1 to 10 The number of attempts to transfer a file. (An attempt is
Default 3 defined as trying to download the file from all IP
addresses that map to a particular domain name.)
Retry Wait 0 to 300 The minimum amount of time that must elapse before
Default 1 retrying a file transfer, in seconds. The time is measured
from the start of a transfer attempt which is defined as the
set of upload/download transactions made with the IP
addresses that map to a given provisioning server's DNS
host name. If the set of transactions in an attempt is equal
to or greater than the Retry Wait value, then there will be
no further delay before the next attempt is started.
For more information, refer to Deploying Phones From the
Provisioning Server on page 3-17.
Tag SN to UA Disabled, Enabled If enabled, the phone’s serial number (MAC address) is
included in the User-Agent header of the Microbrowser.
The default value is Disabled.
Note The Server User and Server Password parameters should be changed from the
default values. Note that for insecure protocols the user chosen should have very
few privileges on the server.
3 - 11
Ethernet Menu
The following Ethernet configuration parameters can be modified on the
Ethernet menu:
LLDP Enabled, Disabled If enabled, the phone will use the LLDP protocol to
communicate with the network switch for certain network
parameters. Most often this will be used to set the VLAN
that the phone should use for voice traffic. It also reports
power management to the switch. The default value is
Enabled.
If the switch does not support it, VLAN Discovery is used.
Refer to DHCP Menu on page 3-8.
There are four ways to get VLAN on the phone and they
can all be turned on, but the VLAN used is chosen by
priority of each method. The priority is: 1. LLDP; 2. CDP;
3. DVD (VLAN Via DHCP); 4. Static (VLAN ID entered in
config menu).
For more information, refer to LLDP and Supported TLVs
on page C-30.
CDP Compatibility Enabled, Disabled If enabled, the phone will use CDP compatible signaling to
communicate with the network switch for certain network
parameters. Most often this will be used to set the VLAN
that the phone should use for Voice Traffic, and for the
phone to communicate its PoE power requirements to the
switch. The default value is Enabled.
VLAN ID Null, 0 through 4094 Phone’s 802.1Q VLAN identifier. The default value is Null.
Note: Null = no VLAN tagging
VLAN Filtering Enabled, Disabled Filter received Ethernet packets so that the TCP/IP stack
does not process bad data or too much data.
Enable/disable the VLAN filtering state.
The default value is Disabled.
3 - 12
Storm Filtering Enabled, Disabled Filter received Ethernet packets so that the TCP/IP stack
does not process bad data or too much data.
Enable/disable the DoS storm prevention state.
The default value is Enabled.
LAN Port Mode 0 = Auto The network speed over the Ethernet.
1 = 10HD The default value is Auto.
2 = 10FD
HD means half duplex and FD means full duplex.
3 = 100HD
4 = 100FD Note: Polycom recommends that you do not change this
5 = 1000FD setting.
PC Port Mode 0 = Auto The network speed over the Ethernet.

1 = 10HD The default value is Auto.
2 = 10FD
HD means half duplex and FD means full duplex.
3 = 100HD
4 = 100FD Note: Polycom recommends that you do not change this
5 = 1000FD setting unless you want to disable the PC port.
-1 = Disabled
Note The LAN Port Mode applies to all phones supported by SIP 3.0 . The PC Port Mode
parameters are only available on phones with a second Ethernet port.
Only the SoundPoint IP 560 and 670 and Polycom VVX 1500 phones supports the
LAN Port Mode and PC Port Mode setting of 1000FD.The 1000BT LAN Clock and
1000BT PC Clock parameters are only available on SoundPoint IP 560 and 670
phones
Syslog Menu
Syslog is a standard for forwarding log messages in an IP network. The term
“syslog” is often used for both the actual syslog protocol, as well as the
application or library sending syslog messages.
The syslog protocol is a very simplistic protocol: the syslog sender sends a
small textual message (less than 1024 bytes) to the syslog receiver. The receiver
is commonly called “syslogd”, “syslog daemon” or “syslog server”. Syslog
messages can be sent through UDP, TCP, or TLS. The data is sent in cleartext.
Syslog is supported by a wide variety of devices and receivers. Because of this,
syslog can be used to integrate log data from many different types of systems
into a central repository.
The syslog protocol is defined in RFC 3164. For more information on syslog,
go to http://www.ietf.org/rfc/rfc3164.txt?number=3164 .
3 - 13
The following syslog configuration parameters can be modified on the Syslog

menu:
Server Address dotted-decimal IP address The syslog server IP address or host name.
OR The default value is NULL.
domain name string
Server Type None=0, The protocol that the phone will use to write to the syslog
UDP=1, server.
TCP=2, If set to “None”, transmission is turned off, but the server
TLS=3 address is preserved.
Facility 0 to 23 A description of what generated the log message. For

more information, refer to section 4.1.1 of RFC 3164.
The default value is 16, which maps to “local 0”.
Render Level 0 to 6 Specifies the lowest class of event that will be rendered to
syslog. It is based on log.render.level and can be a
lower value.
Refer to Basic Logging <level/><change/> and <render/>
on page A-106.
Note: Use left and right arrow keys to change values.
Prepend MAC Enabled, Disabled If enabled, the phone’s MAC address is prepended to the
Address log message sent to the syslog server.
Setting Up the Provisioning Server

The provisioning server can be on the local LAN or anywhere on the Internet.
Multiple provisioning servers can be configured by having the provisioning
server DNS name map to multiple IP addresses. The default number of
provisioning servers is one and the maximum number is eight. The following
protocols are supported for redundant provisioning servers: HTTPS, HTTP,
and FTP. For more information on the protocol used on each platform, refer to
Supported Provisioning Protocols on page 3-4.
All of the provisioning servers must be reachable by the same protocol and the
content available on them must be identical. The parameters described in
section Server Menu on page 3-10 can be used to configure the number of times
each server will be tried for a file transfer and also how long to wait between
each attempt. The maximum number of servers to be tried is configurable. For
more information, contact your Certified Polycom Reseller.
3 - 14
Note Be aware of how logs, overrides and directories are uploaded to servers that map
to multiple IP addresses. The server that these files are uploaded to may change
over time.
If you want to use redundancy for uploads, synchronize the files between servers in
the background.
However, you may want to disable the redundancy for uploads by specifying
specific IP addresses instead of URLs for logs, overrides, and directory in the
<MAC-address>.cfg .
To set up the provisioning server:
Note Use this procedure as a recommendation if this is your first provisioning server
setup.
1. Install a provisioning server application or locate suitable existing

server(s).
Polycom recommends that you use RFC-compliant servers.
2. Create an account and home directory.
Note If the provisioning protocol requires an account name and password, the server
account name and password must match those configured in the phones. Defaults
are: provisioning protocol: FTP, name: PlcmSpIp, password: PlcmSpIp.
Each phone may open multiple connections to the server.

The phone will attempt to upload log files, a configuration override file,
and a directory file to the server. This requires that the phone’s account has
delete, write, and read permissions. The phone will still function without
these permissions, but will not be able to upload files.
The files downloaded from the server by the phone should be made
read-only.
Note Typically all phones are configured with the same server account, but the server
account provides a means of conveniently partitioning the configuration. Give each
account an unique home directory on the server and change the configuration on
an account-by-account basis.
3 - 15
3. Copy all files from the distribution zip file to the phone home directory.
Maintain the same folder hierarchy.
There are two distribution zip files. The combined image file contains:
— sip.ld
— sip.cfg
— phone1.cfg
— 000000000000.cfg
— 000000000000-directory~.xml
— SoundPointIP-dictionary.xml (one of each supported language)
— SoundPointIPWelcome.wav
The split image file contains individual sip.ld files for each model as well
as the configuration files and dictionary files.
Refer to the latest Release Notes for a detailed description of each file in the
distribution and further information on determining which distribution to
use.
Provisioning Server Security Policy

You must decide on a provisioning server security policy.
Polycom recommends allowing file uploads to the provisioning server where the
security environment permits. This allows event log files to be uploaded and
changes made by the phone user to the configuration (through the web server and
local user interface) and changes made to the directory to be backed up. This
greatly eases our ability to provide customer support in diagnosing issues that may
occur with the phone operation.
For organizational purposes, configuring a separate log file directory, override

directory, contact directory, and license directory is recommended, but not
required. The different directories can have different access permissions. For
example, for LOG, CONTACTS, and OVERRIDES, allow full access (read and
write) and for all others, read-only access. For more information on
LOG_FILE_DIRECTORY, OVERRIDES, CONTACTS, and LICENSE, refer to
Master Configuration Files on page A-2.
File permissions should give the minimum access required and the account
used should have no other rights on the server.
The phone's server account needs to be able to add files to which it can write
in the log file directory and the root directory. It must also be able to list files
in all directories mentioned in the <MAC-address>.cfg file. All other files that
the phone needs to read, such as the application executable and the standard
configuration files, should be made read-only through file server file
permissions.
3 - 16
Deploying Phones From the Provisioning Server

You can successfully deploy SoundPoint IP / SoundStation IP / VVX phones
from one or more provisioning servers.
For all SoundPoint IP / SoundStation IP / VVX phones, follow the normal
provisioning process in the next section, Provisioning Phones. However, if you
have decided to daisy-chain two SoundStation IP 7000 conference phones
together, read the information in Provisioning SoundStation IP 7000 Phones
Using C-Link on page 3-20 to understand the different provisioning options
available.
Provisioning Phones
The default configuration files will work without any changes; however, if you
change any configuration file, then the others will have to adjusted
accordingly.
For more information on why to create another configuration file, refer to the
“Configuration File Management on SoundPoint IP, SoundStation IP, and
Polycom VVX 1500 Phones” white paper at
http://www.polycom.com/global/documents/support/technical/products
/voice/white_paper_configuration_file_management_on_soundpoint_ip_ph
ones.pdf .
For more information on phone configuration and provisioning, refer to the
appropriate Technical Bulletins and Quick Tips at
http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical
_Bulletins_pub.html .
For more information on encrypting configuration files, refer to Encrypting
Configuration Files on page C-4.
To deploy phones from the provisioning server:

1. Create per-phone configuration files by performing the following steps:
a Obtain a list of phone Ethernet addresses (barcoded label on
underside of phone and on the outside of the box).
b Create per-phone phone[MACaddress].cfg file by using the
phone1.cfg file from the distribution as templates.
For more information on the phone1.cfg file, refer to Per-Phone
Configuration on page A-133.
Note Throughout this guide, the terms Ethernet address and MAC address are used
interchangeable.
Do not use [MACaddress]-phone.cfg as the per-phone filename. This filename is
used by the phone itself to store user preferences (overrides).
3 - 17
c Edit contents of phone[MACaddress].cfg if desired.

For example, edit the parameters.
2. Create new configuration file(s) in the style of sip.cfg by performing the
following steps:
a Create application sipXXXX.cfg file by using the sip.cfg file from the
distribution as templates.
For more information on the sip.cfg file, refer to Application
b Edit contents of sipXXXX.cfg if desired.
For example, edit the parameters.
Most of the default settings are typically adequate, however, if SNTP
settings are not available through DHCP, the SNTP GMT offset and
(possibly) the SNTP server address will need to be edited for the
correct local conditions. Changing the default daylight savings
parameters will likely be necessary outside of North American
locations. (Optional) Disable the local web (HTTP) server or change its
signaling port if local security policy dictates (refer to Web Server
<httpd/> on page A-79). Change the default location settings for user
interface language and time and date format (refer to Localization
<lcl/> on page A-27).
3. Create a master configuration file by performing the following steps:
a Create per-phone or per-platform <MACaddress>.cfg files by using
the 00000000000.cfg and files from the distribution as templates.
For more information, refer to Master Configuration Files on page
A-2.
b Edit the CONFIG_FILES attribute of the <MACaddress>.cfg files so
that it references the appropriate phone[MACaddress].cfg file.
For example, replace the reference to phone1.cfg with
phone[MACaddress].cfg.
c Edit the CONFIG_FILES attribute of the <MACaddress>.cfg files so
that it references the appropriate sipXXXX.cfg file.
For example, replace the reference to sip.cfg with sip650.cfg.
d Edit the LOG_FILE_DIRECTORY attribute of the <MACaddress>.cfg
files so that it points to the log file directory.
e Edit the CONTACT_DIRECTORY attribute of the
<MACaddress>.cfg files so that it points to the organization’s contact
directory.
3 - 18
4. Reboot the phones by pressing the reboot multiple key combination.

For more information, refer to Multiple Key Combinations on page C-11.
The BootROM and SIP application modify the APPLICATION
APP_FILE_PATH attribute of the <MACaddress>.cfg files so that it
references the appropriate sip.ld files.
For example, the reference to sip.ld is changed to 2345-11670-001.sip.ld to
boot the SoundPoint IP 670 image.
Note At this point, the phone sends a DHCP Discover packet to the DHCP server. This is
found in the Bootstrap Protocol/option "Vendor Class Identifier" section of the
packet and includes the phone’s part number and the BootROM version.
For example, a SoundPoint IP 650 might send the following information:
5EL@
DC?5cSc52*46*(9N7*<u6=pPolycomSoundPointIP-SPIP_6502345-12600-001,1B
R/4.0.0.0155/23-May-07 13:35BR/4.0.0.0155/23-May-07 13:35
For more information, refer to Parsing Vendor ID Information on page C-24.
5. Ensure that the configuration process completed correctly.

For example, on the phone, press the Menu key, and then select Status >
Platform > Application to see the SIP application version and Status >
Platform > Configuration to see the configuration files downloaded to the
phone.
Monitor the provisioning server event log and the uploaded event log files
(if permitted). All configuration files used by the provisioning server are
logged.
You can now instruct your users to start making calls.
3 - 19
Provisioning SoundStation IP 7000 Phones Using C-Link

Normally the SoundStation IP 7000 conference phone is provisioned over the
Ethernet by the provisioning server. However, when two SoundStation IP
7000 phones are daisy-chained together, the one that is not directly connected
to the Ethernet can still be provisioned (known as the secondary).
Power Adapter
Multi-Interface
Module
5
12-foot
Ethernet Cable
Interconnect Cable
25-foot
Network Cable 4
The provisioning over C-Link feature is automatically enabled when a

SoundStation IP 7000 phone is not connected to the Ethernet. Both
SoundStation IP 7000 phones must be running the same version of the SIP
application.
The steps for provisioning the secondary SoundStation IP 7000 phone are the
same as for the primary SoundStation IP 7000 phone. You can reboot the
primary without rebooting the secondary. However, the primary and
secondary should be rebooted together for the primary/secondary
relationship to be recognized. If you power up both SoundStation IP 7000
phones, the primary will power up first.
Currently, provisioning over C-Link is supported for the following
configurations of SoundStation IP 7000 conference phones:
• Two SoundStation IP 7000 conference phone daisy-chained together
• Two SoundStation IP 7000 conference phone daisy-chained together with

one external microphone, specifically designed for the SoundStation IP
7000 conference phone
The provisioning server (or proxy) for the secondary is determined by the
following criteria:
• The primary phone must be powered up using Multi-Interface Module.

PoE will not provide enough power for both phones.
• If the secondary is configured for DHCP, use the primary’s provisioning

server if the primary is configured for DHCP.
3 - 20
• If the secondary is not configured for DHCP, use the secondary’s static
provisioning server if it exists.
• If the secondary’s static provisioning server does not exists, use the
primary’s provisioning server (ignoring the source).
For more information on daisy-chaining and setting up the SoundStation IP
7000 conference phone, refer to the Setup Guide for the Polycom SoundStation IP
7000 Phone, which is available at
http://www.polycom.com/voicedocumentation/.
Upgrading SIP Application

You can upgrade the SIP application that is running on the SoundPoint IP,
SoundStation IP, and VVX phones in your organization. The exact steps that
you perform are dependent on the version of the SIP application that is
currently running on the phones and the version that you want to upgrade to.
The BootROM, application executable, and configuration files can be updated
automatically through the centralized provisioning model. These files are
read-only by default.
Most organization can use the instructions shown in the next section,
Supporting SoundPoint IP, SoundStation IP, and Polycom VVX Phones.
However, if your organization has a mixture of SoundPoint IP 300, 301, 500,
501, 600, 601 and/or SoundStation IP 4000 phones deployed along with other
models, you will need to change the phone configuration files to continue to
support the SoundPoint IP 300, 301, 500, 501, 600, and 601 and SoundStation
IP 4000 phones when software releases SIP 3.2.0 or later are deployed. These
models were discontinued as follows:
• The SoundPoint IP 300 and 500 phones as of May 2006.
• The SoundPoint IP 301, 600, and 601 phones as March 2008.
• The SoundPoint IP 501 phone as of August 2009.
• The SoundStation IP 4000 phone as of May 2009.

In all cases, refer to Supporting SoundPoint IP 300, 301, 500, 501, 600 and 601
and SoundStation IP 4000 Phones on page 3-23.
3 - 21
Warning The SoundPoint IP 300 and 500 phones will be supported on the latest
maintenance patch release of the SIP 2.1 software stream—currently SIP 2.1.4.
Any critical issues that affect SoundPoint IP 300 and 500 phones will be addressed
by a maintenance patch on this stream until the End of Life date for these products.
Phones should be upgraded to BootROM 4.0.0 for these changes to be effective.
The SoundPoint IP 301, 501, 600, and 601 and the SoundStation IP 4000 phones
will be supported on the latest maintenance patch release of the SIP 3.1 software
stream—currently SIP 3.1.3 . Any critical issues that affect SoundPoint IP 300 and
500 phones will be addressed by a maintenance patch on this stream until the End
of Life date for these products. Phones should be upgraded to BootROM 4.0.0 or
later for these changes to be effective.
Supporting SoundPoint IP, SoundStation IP, and Polycom VVX Phones
Warning If you need to upgrade any Polycom VVX 1500 phones running SIP 3.1.3 or earlier
to SIP 3.2.2, you must perform additional steps before rebooting the phone to
download the new SIP software. Refer to “Technical Bulletin 53522: Upgrading the
Polycom VVX 1500 Phone to SIP 3.2.2” at
tins_pub.html .
To automatically update:
1. Back up old application and configuration files.
The old configuration can be easily restored by reverting to the backup
files.
2. Customize new configuration files or apply new or changed parameters
to the old configuration files.
Differences between old and new versions of configuration files are
explained in the Release Notes that accompany the software. Both
mandatory and optional changes may present. Changes to site-wide
configuration files such as sip.cfg can be done manually, but a scripting
tool is useful to change per-phone configuration files.
Warning The configuration files listed in CONFIG_FILES attribute of the master configuration
file must be updated when the software is updated. Any new configuration files
must be added to the CONFIG_FILES attribute in the appropriate order.
Mandatory changes must be made or the software may not behave as expected.
For more information, refer to the “Configuration File Management on SoundPoint
IP, SoundStation IP, and Polycom VVX 1500 Phones” white paper at
http://www.polycom.com/global/documents/support/technical/products/voice/white_
paper_configuration_file_management_on_soundpoint_ip_phones.pdf .
3. Save the new configuration files and images (such as sip.ld) on the
3 - 22
4. Reboot the phones using automatic methods such as polling or

check-sync.
Using the reboot multiple key combination should be done as a backup
option only. For more information, refer to Multiple Key Combinations on
page C-11.
Since the APPLICATION APP_FILE_PATH attribute of the
<MACaddress>.cfg files references the individual sip.ld files, it is
possible to verify that an update is applied to phones of a particular
model.
For example, the reference to sip.ld is changed to 2345-11670-001.sip.ld to
boot the SoundPoint IP 670 image.
The phones can be rebooted remotely through the SIP signaling protocol.
Refer to Special Events <specialEvent/> on page A-20.
The phones can be configured to periodically poll the provisioning server to
check for changed configuration files or application executable. If a change is
detected, the phone will reboot to download the change. Refer to Provisioning
<prov/> on page A-114.
Supporting SoundPoint IP 300, 301, 500, 501, 600 and 601 and
SoundStation IP 4000 Phones
With enhancements available since BootROM 4.0.0 and SIP 2.1.2, you can
modify the 000000000000.cfg or <MACaddress>.cfg configuration file to
direct phones to load the software image and configuration files based on the
phone model number. Refer to Master Configuration Files on page A-2.
The SIP 3.2.0 or later software distributions contain only the new distribution
files for the new release. You must rename the sip.ld, sip.cfg, and phone1.cfg
from a previous 2.1.x distribution that is compatible with SoundPoint IP 300
and 500 phones or a previous 3.1.y distribution that is compatible with
SoundPoint IP 301, 501, 600, and 601 SoundStation IP 4000 phones.
The following procedure must be used for upgrading to SIP 3.2.0 or later for
installations that have SoundPoint IP 300, 301, 500, 501, 600, 601 and
SoundStation IP 4000 phones deployed. It is also recommended that this same
approach be followed even if these phones are not part of the deployment as
it will simplify management of phone systems with future software releases.
To upgrade your SIP application:

1. Do one of the following steps:
a Place all bootrom.ld files corresponding to BootROM release zip file
onto the provisioning server.
b Ensure that all phones are running BootROM 4.0.0 or later code.
3 - 23
2. Copy sip.ld, sip.cfg and phone1.cfg from the SIP 3.2.0 or later release
distribution onto the provisioning server.
These are the relevant files for all phones except the SoundPoint IP 300,
301, 500, 501, 600, 601 and SoundStation IP 4000 phones.
3. Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to
sip_21x.ld, sip_21x.cfg, and phone1_21x.cfg respectively on the
These are the relevant files for supporting the SoundPoint IP 300 and 500
phones.
4. Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to
sip_31y.ld, sip_31y.cfg, and phone1_31y.cfg respectively on the
These are the relevant files for supporting the SoundPoint IP 301, 501, 600,
601 and SoundStation IP 4000 phones.
5. Modify the 000000000000.cfg file, if required, to match your configuration
file structure.
For example:
<APPLICATION
APP_FILE_PATH="sip.ld"
APP_FILE_PATH_SPIP500="sip_214.ld"
APP_FILE_PATH_SSIP4000="sip_313.ld"
CONFIG_FILES="[PHONE_MAC_ADDRESS]-user.cfg, phone1.cfg, sip.cfg"
CONFIG_FILES_SPIP500="[PHONE_MAC_ADDRESS]-user.cfg,
phone1_214.cfg, sip_214.cfg"
CONFIG_FILES_SSIP4000="[PHONE_MAC_ADDRESS]-user.cfg,
MISC_FILES=""
LOG_FILE_DIRECTORY=""
OVERRIDES_DIRECTORY=""
CONTACTS_DIRECTORY=""
/>
3 - 24
6. Remove any <MACaddress>.cfg files that may have been used with
earlier releases from the provisioning server.
Note This approach takes advantage of an enhancement that was added in

SIP2.0.1/BootROM 3.2.1 that allows for the substitution of the phone specific
[MACADDRESS] inside configuration files. This avoids the need to create unique
<MACaddress>.cfg files for each phone such that the default 000000000000.cfg
file can be used for all phones in a deployment.
If this approach is not used, then changes will need to be made to all the
<MACaddress>.cfg files for SoundPoint IP 300, 301, 500, 501, 600, and 601 and
SoundStation IP 4000 phones or all of the <MACaddress>.cfg files if it is not
explicitly known which phones are SoundPoint IP 300 and 500 phones.
For more information, refer to “Technical Bulletin 35311: Supporting

SoundPoint IP 300, 301, 500, 501, 600, and 601 and SoundStation IP 4000
Phones with SIP 2.2.0 or SIP 3.2.0 and Later Releases“at
3 - 25
3 - 26
4
Configuring Your System
After you set up your SoundPoint IP / SoundStation IP / VVX phones on the

network, you can allow users to place and answer calls using the default
configuration, however, you may require some basic changes to optimize your
system for best results.
This chapter provides information for making configuration changes for:
• Setting Up Basic Features
• Setting Up Advanced Features
• Setting Up Audio Features
• Setting Up Video Features
• Setting Up Security Features

This chapter also provides instructions on:
• Configuring SoundPoint IP / SoundStation IP / VVX Phones Locally

To troubleshoot any problems with your SoundPoint IP / SoundStation IP /
VVX phones on the network, refer to Troubleshooting Your SoundPoint IP /
SoundStation IP / VVX Phones on page 5-1. For more information on the
configuration files, refer to Configuration Files on page A-1.
Setting Up Basic Features

This section provides information for making configuration changes for the
following basic features:
• Call Log
• Call Timer
• Call Waiting
• Called Party Identification
• Calling Party Identification
4-1
• Missed Call Notification
• Connected Party Identification
• Context Sensitive Volume Control
• Customizable Audio Sound Effects
• Message Waiting Indication
• Distinctive Incoming Call Treatment
• Distinctive Ringing
• Distinctive Call Waiting
• Do Not Disturb
• Handset, Headset, and Speakerphone
• Local Contact Directory
• Local Digit Map
• Microphone Mute
• Soft Key Activated User Interface
• Speed Dial
• Time and Date Display
• Idle Display Animation
• Ethernet Switch
• Graphic Display Backgrounds

This section also provides information for making configuration changes for
the following basic call management features:
• Automatic Off-Hook Call Placement
• Call Hold
• Call Transfer
• Local / Centralized Conferencing
• Call Forward
• Directed Call Pick-Up
• Group Call Pick-Up
• Call Park/Retrieve
• Last Call Return
4-2
Call Log
The phone maintains a call log. The log contains call information such as
remote party identification, time and date, and call duration. It can be used to
redial previous outgoing calls, return incoming calls, and save contact
information from call log entries to the contact directory.
The call log is stored in volatile memory and is maintained automatically by
the phone in three separate lists: Missed Calls, Received Calls and Placed
Calls. The call lists can be cleared manually by the user and will be erased
when the phone is restarted.
Note On some SoundPoint IP platforms, missed calls and received calls appear in one
list. Missed calls appear as and received calls appear as .

The “call list” feature can be disabled on all SoundPoint IP and SoundStation IP
platforms except the SoundPoint IP 32x/33x and SoundStation IP 7000.
Configuration changes can be performed centrally at the provisioning server:
Central Configuration File: Enable or disable all call lists or individual call lists.
(provisioning sip.cfg • For more information, refer to Feature <feature/> on page A-116.
server)
Call Timer
A call timer is provided on the display. A separate call timer is maintained for
each distinct call in progress. The call duration appears in hours, minutes, and
seconds.
There are no related configuration changes.
Call Waiting
When an incoming call arrives while the user is active on another call, the
incoming call is presented to the user visually on the LCD display. A
configurable sound effect such as the familiar call-waiting beep will be mixed
with the active call audio as well.
Configuration changes can performed centrally at the provisioning server:
Central Configuration File: Specify the ring tone heard on an incoming call when another call is
(provisioning phone1.cfg active.
server) • For more information, refer to Call Waiting <callWaiting/> on page
A-142.
Disable call waiting.
• For more information, refer to Registration <reg/> on page A-134.
4-3
For related configuration changes, refer to Customizable Audio Sound Effects

on page 4-6.
Called Party Identification

The phone displays and logs the identity of the remote party specified for
outgoing calls. This is the party that the user intends to connect with.
The identity displayed is based on the number of the placed call and
information obtained from the network signaling.
Note The phone does not match the number of the placed call to any entries in the Local
Contact Directory or Corporate Directory.
Calling Party Identification

The phone displays the caller identity, derived from the network signaling,
when an incoming call is presented, if the information is provided by the call
server. For calls from parties for which a directory entry exists, the local name
assigned to the Contact Directory entry may optionally be substituted.
Note The phone does not match the received number to any entries in the Corporate
Directory.
Configuration changes can performed centrally at the provisioning server or

locally:
Central Configuration File: Specify whether or not to use directory name substitution.
(provisioning sip.cfg • For more information, refer to User Preferences <up/> on page
server) A-31.
Local Web Server Specify whether or not to use directory name substitution.
(if enabled) Navigate to: http://<phoneIPAddress>/coreConf.htm#us
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the provisioning server. Changes will
permanently override global settings unless deleted through the
Reset Local Config menu selection.
4-4
Missed Call Notification

The phone can display the number of calls missed since the user last looked at
the Missed Calls list. The phone can be configured to use a built-in missed call
counter or to display information provided by a Session Initiation Protocol
(SIP) server.
Note On some SoundPoint IP platforms, missed calls and received calls appear in one
list.
Central Configuration file: Turn this feature on or off.

server)
Configuration file: Specify per-registration whether all missed-call events or only
phone1.cfg remote/server-generated missed-call events will be displayed.
• For more information, refer to Missed Call Configuration
<serverMissedCall/> on page A-141.
Connected Party Identification

The identity of the remote party to which the user has connected is displayed
and logged, if the name and ID is provided by the call server. The connected
party identity is derived from the network signaling. In some cases the remote
party will be different from the called party identity due to network call
diversion. For example, Bob places a call to Alice, but he ends up connected to
Fred.
Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive
volume of call audio is adjustable for speakerphone, handset, and headset
separately. While transmit levels are fixed according to the TIA/EIA-810-A
standard, receive volume is adjustable. For SoundPoint IP and VVX phones, if
using the default configuration parameters, the receive handset/headset
volume resets to nominal after each call to comply with regulatory
requirements. Handsfree volume persists with subsequent calls.
Central Configuration file: Adjust receive and handset/headset volume.

(provisioning sip.cfg • For more information, refer to Volume Persistence <volume/> on
server) page A-50.
4-5
Customizable Audio Sound Effects

Audio sound effects used for incoming call alerting and other indications are
customizable. Sound effects can be composed of patterns of synthesized tones
or sample audio files. The default sample audio files may be replaced with
alternates in .wav file format. Supported .wav formats include:
• mono G.711 (13-bit dynamic range, 8-khz sample rate)
• mono L16/16000 (16-bit dynamic range, 16-kHz sample rate)
Note L16/32000 and L16/48000 are only supported on SoundPoint IP 7000 phones.
Note The alternate sampled audio sound effect files must be present on the provisioning
server or the Internet for downloading at boot time.
Configuration changes can be performed centrally at the provisioning server

or locally:
Central Configuration File: Specify patterns used for sound effects and the individual tones or
(provisioning sip.cfg sampled audio files used within them.
server) • For more information, refer to Sampled Audio for Sound Effects
<saf/> on page A-37 or Sound Effects <se/> on page A-38.
Local Web Server Specify sampled audio wave files to replace the built-in defaults.
(if enabled) Navigate to http://<phoneIPAddress>/coreConf.htm#sa
4-6
Message Waiting Indication

The phone will flash a message-waiting indicator (MWI) LED when instant
messages and voice messages are waiting.
Central Configuration file: Specify per-registration whether the MWI LED is enabled or disabled.
(provisioning phone1.cfg • For more information, refer to Message Waiting Indicator <mwi/>
server) on page A-149.
Specify whether MWI notification is displayed for registration x
(pre-SIP 2.1 behavior is enabled).
• For more information, refer to User Preferences <up/> on page
A-31.
Distinctive Incoming Call Treatment

The phone can automatically apply distinctive treatment to calls containing
specific attributes. The distinctive treatment that can be applied includes
customizable alerting sound effects and automatic call diversion or rejection.
Call attributes that can trigger distinctive treatment include the calling party
name or SIP contact (number or URL format).
For related configuration changes, refer to Local Contact Directory on page
4-10.
Distinctive Ringing
There are three options for distinctive ringing:
1. The user can select the ring type for each line by pressing the Menu key,
and then selecting Settings > Basic > Ring Type. This option has the
third (lowest) priority.
2. The ring type for specific callers can be assigned in the contact directory.
For more information, refer to Distinctive Incoming Call Treatment, the
previous section. This option is second in priority.
3. The voIpProt.SIP.alertInfo.x.value and
voIpProt.SIP.alertInfo.x.class fields can be used to map calls to
specific ring types. This option requires server support and is first
(highest) in priority.
4-7

or locally:
Central Configuration file: Specify the mapping of Alert-Info strings to ring types.
(provisioning sip.cfg • For more information, refer to Alert Information <alertInfo/> on
server) page A-19.
Configuration file: Specify the ring type to be used for each line.
phone1.cfg • For more information, refer to Registration <reg/> on page A-134.
XML File: <Ethernet This file can be created manually using an XML editor.
address>-directory. • For more information, refer to Local Contact Directory on page
xml 4-10.
Local Local Phone User The user can edit the ring types selected for each line under the
Interface Settings menu. The user can also edit the directory contents.
Reset Local Config menu selection and the <Ethernet
address>-phone.cfg is removed from the provisioning server.
Distinctive Call Waiting

The voIpProt.SIP.alertInfo.x.value and
voIpProt.SIP.alertInfo.x.class fields can be used to map calls to distinct
call waiting types, currently limited to two styles. This feature requires server
support.
Central Configuration file: Specify the mapping of Alert-Info strings to call waiting types.
(provisioning sip.cfg • For more information, refer to Alert Information <alertInfo/> on
server) page A-19.
Do Not Disturb
A Do Not Disturb (DND) feature is available to temporarily stop all incoming
call alerting. Calls can optionally be treated as though the phone is busy while
DND is enabled. DND can be configured as a per-registration feature.
Incoming calls received while DND is enabled are logged as missed. For more
information on forwarding calls while DND is enabled, refer to Call Forward
on page 4-22.
Server-based DND is active if the feature is enabled on both the phone and the
server and the phone is registered. The server-based DND feature is applicable
for all registrations on the phone (no per-registration mode) and it disables
local Call Forward and DND features unless configured otherwise.
4-8
Server-based DND will behave the same as per-SIP 2.1 per-registration feature
with the following exceptions:
• Server based DND cannot be used if the phone is configured as a shared

line.
• If server-based DND is enabled, but inactive, and the user presses the
DND key or selects the DND option on the Feature menu, the “Do Not
Disturb” message does not appear on the user’s phone (incoming call
alerting will continue).
or locally:
Central Configuration file: Enable or disable server-based DND.

(provisioning sip.cfg • For more information, refer to SIP <SIP/> on page A-11
server) Enable or disable local DND behavior when server-based enabled.
• For more information, refer to SIP <SIP/> on page A-11.
Specify whether or not DND results in incoming calls being given
busy treatment.
• For more information, refer to Call Handling Configuration <call/>
on page A-80.
Configuration file: Enable or disable server-based DND as a per-registration feature.

phone1.cfg • For more information, refer to Registration <reg/>on page A-134.
Specify whether DND is treated as a per-registration feature or a
global feature on the phone.
• For more information, refer to Do Not Disturb <dnd/> on page
A-144.
Local Local Phone User Enable or disable DND using the Do Not Disturb key on the
Interface SoundPoint IP 550, 560, 650, and 670 and the Polycom VVX 1500 or
the “Do Not Disturb” option on the Features menu on the SoundPoint
IP 32x/33x, 430, and 450 and SoundStation IP 5000, 6000 and 7000.
Note: The LED on the Do Not Disturb key on the Polycom VVX
1500 is red when pressed or when server-based DND is enabled.
Handset, Headset, and Speakerphone

SoundPoint IP phones come standard with a handset and a dedicated
connector is provided for a headset (not supplied). All Polycom phones are
full-duplex speakerphones. The SoundPoint IP phones provide dedicated
keys for convenient selection of either the speakerphone or headset.
All Polycom desktop phones can be configured to use the electronic
hookswitch. For more information, refer to “Technical Bulletin 35150: Using an
Electronic Hookswitch with SoundPoint IP and Polycom VVX 1500 Phones“at
4-9

or locally:
Central Configuration file: Enable or disable persistent headset mode.

(provisioning sip.cfg For more information, refer to User Preferences <up/> on page A-31.
server) Enable or disable hands-free speakerphone mode.
A-31.
Configuration file: Specify whether or not the electronic hookswitch is enabled and what
phone1.cfg type of headset is attached.
• For more information, refer to User Preferences <up/>on page
A-134.
Local Web Server Enable or disable persistent headset mode.

(if enabled) Navigate to: http://<phoneIPAddress>/coreConf.htm#us
Local Phone User Enable or disable persistent headset mode through the Settings
Interface menu (Settings > Basic > Preferences > Headset > Headset
Memory Mode).
Enable or disable hands-free speakerphone mode through the
Settings menu (Settings > Advanced > Admin Settings > Phone
Settings).
Local Contact Directory

The phone maintains a local contact directory. The directory can be
downloaded from the provisioning server and edited locally (if configured in
that way). Contact information from previous calls may be easily added to the
directory for convenient future access.
The directory is the central database for several other features including
speed-dial, distinctive incoming call treatment, presence, and instant
messaging. The maximum number of entries in the local contact directory is
phone-dependent.
Note If a user makes a change to the local contact directory, there is a five second
timeout before it is uploaded to the provisioning server as
<mac-address>-directory.cfg.
If so configured, the first and last name fields of the local contact directory entries
which match incoming calls will be used for caller identification display and in the
call lists (instead of the name provided through network signaling).
4 - 10

or locally:
Central Configuration file: Set whether the directory uses volatile storage on the phone.
(provisioning sip.cfg • For more information, refer to Local Directory <local/> on page
server) A-86.
Specify whether or not the local contact directory is read only.
• For more information, refer to Local Directory <local/> on page
A-86.
XML file: A sample file named 000000000000-directory~.xml (Note the extra

000000000000-direct “~” in the filename) is included with the application file distribution.
ory.xml This file can be used as a template for the per-phone <Ethernet
address>-directory.xml directories (edit contents, then rename to
<Ethernet address>-directory.xml). It also can be used to seed new
phones with an initial directory (edit contents, then remove “~” from
file name). Telephones without a local directory, such as new units
from the factory, will download the 00000000000-directory.xml
directory and base their initial directory on it. These files should be
edited with an XML editor. These files can be downloaded once per
reflash.
For information on file format, refer to the next section, Local Contact
Directory File Format.
XML file: <Ethernet This file can be created manually using an XML editor.
address>-directory. For information on file format, refer to the next section, Local Contact
xml Directory File Format.
Local Local Phone User The user can edit the directory contents if configured in that way.
Interface Changes will be stored in the phone’s flash file system and backed up
to the provisioning server copy of <Ethernet
address>-directory.xml if this is configured. When the phone boots,
the provisioning server copy of the directory, if present, will overwrite
the local copy.
Local Contact Directory File Format

An example of a local contact directory is shown below. The subsequent table
provides an explanation of each element. Elements can appear in any order.
<?xml version=”1.0” encoding=”UTF-8” standalone=”yes” ?>
<directory>
<item_list>
<item>
<ln>Doe</ln>
<fn>John</fn>
<ct>1001</ct>
<sd>1</sd>
<lb>Mr</lb>
<pt>H323</pt>
<rt>1</rt>
4 - 11
<dc/>
<ad>0</ad>
<ar>0</ar>
<bw>0</bw>
<bb>0</bb>
</item>
...
<item>
<ln>Smith</ln>
<fn>Bill</fn>
<ct>1003</ct>
<sd>3</sd>
<lb>Dr</lb>
<pt>SIP</pt>
<rt>3</rt>
<dc/>
<ad>0</ad>
<ar>0</ar>
<bw>0</bw>
<bb>0</bb>
</item>
</item_list>
</directory>
Element Permitted Values Interpretation
fn UTF-8 encoded string first name

of up to 40 bytes Note: In some cases, this will be less than 40 characters due to
UTF-8’s variable length encoding.
ln UTF-8 encoded string last name

ct UTF-8 encoded string contact

containing digits (the Used by the phone to address a remote party in the same way that a
user part of a SIP string of digits or a SIP URL are dialed manually by the user. This
URL) or a string that element is also used to associate incoming callers with a particular
constitutes a valid SIP directory entry. For Polycom VVX 1500 phones, the maximum field
URL length is 128 characters; for all other phones, the maximum is 32
characters.
Note: This field cannot be null or duplicated.
sd Null, 1 to 9999 speed-dial index

Associates a particular entry with a speed dial bin for one-touch
dialing or dialing from the speed dial menu.
Note: On the SoundPoint IP 32x/33x and the SoundStation IP 6000
and 7000, the maximum speed-dial index is 99.
4 - 12
Element Permitted Values Interpretation
lb UTF-8 encoded string label

Note: The label of a contact directory item is by default the label
attribute of the item. If the label attribute does not exist or is Null, then
the concatenation of first name and last name will be used as label. A
space is added between first and last names.
pt “SIP”, “H323”, or protocol

“Unspecified” The protocol to use when placing a call to this contact.
rt Null, 1 to 21 ring type

When incoming calls can be associated with a directory entry by
matching the address fields, this field is used to specify ring type to
be used.
dc UTF-8 encoded string divert contact

containing digits (the The forward-to address for the autodivert feature.
user part of a SIP
URL) or a string that
constitutes a valid SIP
URL
ad 0,1 auto divert

If set to 1, automatically diverts callers that match the directory entry
to the address specified in divert contact.
Note: If auto-divert is enabled, it has precedence over auto-reject.
ar 0,1 auto-reject
If set to 1, automatically rejects callers that match the directory entry.
Note: If auto-divert is also enabled, it has precedence over
auto-reject.
bw 0,1 buddy watching

If set to 1, add this contact to the list of watched phones.
bb 0,1 buddy block

If set to 1, block this contact from watching this phone.
Local Digit Map

The phone has a local digit map feature to automate the setup phase of
number-only calls. When properly configured, this feature eliminates the need
for using the Dial or Send soft key when making outgoing calls. As soon as a
digit pattern matching the digit map is found, the call setup process will
complete automatically. The configuration syntax is based on
recommendations in 2.1.5 of RFC 3435. The phone behavior when the user
dials digits that do not match the digit map is configurable. It is possible to
strip a trailing # from the digits sent or to replace certain matched digits (with
4 - 13
the introduction of “R” to the digit map). It is also possible to direct the
protocol used to place a call (with the introduction of “S” and “H” to the digit
map).
For more information digit maps, refer to “Technical Bulletin 11572: Changes
to Local Digit Maps on SoundPoint IP / SoundStation IP / VVX Phones“ at
Note Digit maps do not apply to on-hook dialing. The parameter settings described in
Dial Plan <dialplan/> on page A-23 are ignored during on-hook dialing.

or locally:
Central Configuration file: Specify impossible match behavior, trailing # behavior, digit map
(provisioning sip.cfg matching strings, and time out value.
server) • For more information, refer to Dial Plan <dialplan/> on page A-23.
Configuration file: Specify per-registration impossible match behavior, trailing #

phone1.cfg behavior, digit map matching strings, and time out values that
override those in sip.cfg.
• For more information, refer to Dial Plan <dialplan/> on page
A-145.
Local Web Server Specify impossible match behavior, trailing # behavior, digit map
(if enabled) matching strings, and time out value.
Navigate to: http://<phoneIPAddress>/appConf.htm#ls
Microphone Mute
A microphone mute feature is provided. When activated, visual feedback is
provided. This is a local function and cannot be overridden by the network.
Soft Key Activated User Interface

The user interface makes extensive use of intuitive, context-sensitive soft key
menus. The soft key function is shown above the key on the graphic display.
4 - 14
Using the Configurable Soft Key configuration parameters, an administrator

can modify the default soft keys by removing them at different call stages
and/or adding specific single or multiple functions. Refer to Enhanced
Feature Keys on page 4-38 and Configurable Soft Keys on page 4-48.
Speed Dial
Entries in the local directory can be linked to the speed dial system. The speed
dial system allows calls to be placed quickly from dedicated keys as well as
from a speed dial menu.
For SoundPoint IP 32x/33x desktop phones and SoundStation IP 6000 and
7000 conference phones, the speed dial index range is 1 to 99. For all other
SoundPoint IP and Polycom VVX phones, the range is 1 to 9999.
If Presence watching is enabled for speed dial entries, their status will be
shown on the idle display (if the SIP server supports this feature). For more
information, refer to Presence on page 4-64.
Configuration changes can performed centrally at the provisioning server or
locally:
Central XML file: The <sd>x</sd> element in the <Ethernet address>-directory.xml

(provisioning <Ethernet file links a directory entry to a speed dial resource within the phone.
server) address>-directory. Speed dial entries are mapped automatically to unused line keys (line
xml keys are not available on the SoundStation IP 6000 and 7000) and
are available for selection within the speed dial menu. (Press the Up
arrow key from the idle display to jump to the Speed Dial list).
• For more information, refer to Local Contact Directory on page
4-10.
Local Local Phone User The next available Speed Dial Index is assigned to new directory
Interface entries. Key pad short cuts are available to facilitate assigning and
modifying the Speed Dial Index value for entries in the directory. The
Speed Dial Index field is used to link directory entries to speed dial
operations.
Changes will be stored in the phone’s flash file system and backed up
the local copy.
Time and Date Display

The phone maintains a local clock and calendar. Time and date can be
displayed in certain operating modes such as when the phone is idle and
during a call. The clock and calendar must be synchronized to a remote Simple
Network Time Protocol (SNTP) timeserver. The time and date displayed on
the phone will flash continuously to indicate that they are not accurate until a
successful SNTP response is received. The time and date display can use one
4 - 15
of several different formats and can be turned off. The SoundPoint IP 32x/33x
and IP 4xx phones have a limited selection of date formats due to a smaller
display size.
or locally:
Central Configuration file: Turn time and date display on or off.

server) A-31.
Set the time and date display formats.
• For more information, refer to Date and Time <datetime/> on page
A-31.
Set the basic SNTP settings and daylight savings parameters.
• For more information, refer to Time Synchronization <sntp/> on
page A-75.
Local Web Server Set the basic SNTP and daylight savings settings.
(if enabled) Navigate to: http://<phoneIPAddress>/coreConf.htm#ti
Local Phone User The basic SNTP settings can be made in the Network Configuration
Interface menu.
For more information, refer to DHCP or Manual TCP/IP Setup on
page 3-2.
The user can edit the time and date format and enable or disable the
time and date display under the Settings menu.
address>-phone.cfg on the provisioning server. They will permanently
override global settings unless deleted through the Reset Local
Config menu selection.
Idle Display Animation

All phones can display a customized animation on the idle display in addition
to the time and date. For example, a company logo could be displayed (refer
to Adding a Background Logo on page C-6).
Note Currently customized animations are not supported on the Polycom VVX 1500.
4 - 16
Central Configuration file: To turn idle display animation on or off.

(provisioning sip.cfg • For more information, refer to Indicators <ind/> on page A-100.
server) To replace the animation used for the idle display.
• For more information, refer to Animations <anim/> <IP_330/>,
<IP_335/>, <IP_400/>, <IP_450/>, <IP_600/>, <IP_4000/>, and
<IP_7000/> on page A-101.
To change the position of the idle display animation.
• For more information, refer to Graphic Icons <gi/> <IP_330>,
<IP_400/>, <IP_450/>, <IP_600/>, <IP_4000/>, and <IP_7000/>
on page A-103.
Ethernet Switch
The SoundPoint IP phones (except the SoundPoint IP 32x) and the Polycom
VVX 1500 contain two Ethernet ports, labeled LAN and PC, and an embedded
Ethernet switch that runs at full line-rate. The SoundStation IP phones contain
only one Ethernet port, labeled LAN. The Ethernet switch allows a personal
computer and other Ethernet devices to connect to the office LAN by daisy
chaining through the phone, eliminating the need for a stand-alone hub. The
SoundPoint IP switch gives higher transmit priority to packets originating in
the phone. The phone can be powered through a local AC power adapter or
can be line-powered (power supplied through the signaling or idle pairs of the
LAN Ethernet cable). Line powering typically requires that the phone plugs
directly into a dedicated LAN jack. Devices that do not require LAN power
can then plug into the SoundPoint IP PC Ethernet port. To disable the PC
Ethernet port, refer to Disabling PC Ethernet Port on page C-27.
SoundPoint IP Switch - Port Priorities

To help ensure good voice quality, the Ethernet switch embedded in the
SoundPoint IP phones should be configured to give voice traffic emanating
from the phone higher transmit priority than those from a device connected to
the PC port. If not using a VLAN (VLAN set to blank in the setup menu), this
will automatically be the case. If using a VLAN, ensure that the 802.1p
priorities for both default and real-time transport protocol (RTP) packet types
are set to 2 or greater. Otherwise, these packets will compete equally with
those from the PC port. For more information, refer toVoice Settings <voice/>
on page A-44 and Video Settings <video/> on page A-64.
Graphic Display Backgrounds

You can set up a picture or design to be displayed on the background of the
graphic display of all SoundPoint IP 450, 550, 560, 650, and 670 and Polycom
VVX 1500 phones.
4 - 17
Note When installing a background of your choice, care needs to be taken to ensure that
the background does not adversely affect the visibility of the text on the phone
display. As a general rule, backgrounds should be light in shading for better
usability.
For SoundPoint IP 450, 550, 560, 650, and 670 phones:
• There are a number of default backgrounds, both solid color and pictures.
Both BMP and JPEG files are supported. You can also select the label color
for soft key and line key labels. Users can select which background and
label color appears on their phone.
You can modify the supported solid color and pictures backgrounds. For
example, you can add a gray solid color background or modify a picture
to one of your choice.
For Polycom VVX 1500 phones:
• You can select the pictures or designs displayed on the background. The
supported formats include JPEG, BMP, and PNG and the maximum size
is 800x480. A default picture is displayed when the phone starts up the
first time.
Users can select which background appears on their individual phones.
Users can also select a background from an image displayed by the digital
picture frame feature (refer to Digital Picture Frame on page 4-37).
Note Support for resolutions greater than 800x480 is inconsistent. Content may be
truncated or nor displayed. Progressive/multiscan JPEG images are not supported
at this time.

or locally:
Central Configuration file: Specify which background will be displayed.

(provisioning phone1.cfg • For more information, refer to Backgrounds <bg/> on page A-95.
server)
Local Local Phone User On the Polycom VVX 1500, the user can save one of the images as
Interface the background by selecting Save as Background on the touch
screen.
To modify the backgrounds displayed on the supported SoundPoint IP phones:

1. Modify the sip.cfg configuration file as follows:
a Open sip.cfg in an XML editor.
b Locate the background parameter.
4 - 18
c For the solid backgrounds, set the name and RGB values. For example:
bg.hiRes.gray.pat.solid.3.name=”Gray”
bg.hiRes.gray.pat.solid.3.red=”128”
bg.hiRes.gray.pat.solid.3.green=”128”
bg.hiRes.gray.pat.solid.3.blue=”128”
d For images, select a filename. For example:

bg.hiRes.gray.bm.3.name=”polycom.jpg”
bg.hiRes.gray.bm.3.em.name=”polycomEM.jpg”
bg.hiRes.gray.bm.3.adj=”0”
The default size for images on a phone is 320 x 160. The default size for
images on an Expansion Module is 160 x 320. Use a photo editor on a
computer to adjust the image you want to display. (Edit the image so
the main subject is centered in the upper right corner of the display.)
Download the file to the provisioning server.
e Save the modified sip.cfg configuration file.
Automatic Off-Hook Call Placement

The phone supports an optional automatic off-hook call placement feature for
each registration. This feature is sometimes referred to as ‘hot-dialing’.
Central Configuration file: Specify which registrations have the feature and what contact to call
(provisioning phone1.cfg when going off hook.
server) • For more information, refer to Automatic Off-Hook Call Placement
<autoOffHook/> on page A-140.
Call Hold
The purpose of hold is to pause activity on one call so that the user may use
the phone for another task, such as to make or receive another call. Network
signaling is employed to request that the remote party stop sending media and
to inform them that they are being held. A configurable local hold reminder
feature can be used to remind the user that they have placed calls on hold. The
call hold reminder is always played through the speakerphone.
As of SIP 3.1, you can supply a Music on Hold URI if supported by the call
server. For more information, refer to draft RFC draft-worley-service-example.
4 - 19

or locally:
Central Configuration file: Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
(provisioning sip.cfg a=inactive) outgoing hold signaling is used.
server) • For more information, refer to SIP <SIP/> on page A-11.
Specify local hold reminder options.
• For more information, refer to Hold, Local Reminder
<hold/><localReminder/> on page A-85.
Specify the Music on Hold URI.
• For more information, refer to Music on Hold <musicOnHold/> on
page A-21.
Configuration file: Specify the Music on Hold URI.

phone1.cfg • For more information, refer to Music on Hold <musicOnHold/> on
page A-21.
Local Web Server Specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold
(if enabled) signaling. The alternative is RFC 3264 (a=sendonly or a=inactive).
Navigate to: http://<phoneIPAddress>/appConf.htm#ls
Local Phone User Use the Call Server Configuration menu to specify whether or not to
Interface use RFC 2543 (c=0.0.0.0) outgoing hold signaling. The alternative is
RFC 3264 (a=sendonly or a=inactive).
Call Transfer
Call transfer enables the user (party A) to move an existing call (party B) into
a new call between party B and another user (party C) selected by party A. The
phone offers three types of transfers:
• Blind transfers—The call is transferred immediately to party C after party

A has finished dialing party C’s number. Party A does not hear ring-back.
• Attended transfers—Party A dials party C’s number and hears ring-back

and decides to complete the transfer before party C answers. This option
can be disabled.
• Consultative transfers—Party A dials party C’s number and talks

privately with party C after the call is answered, and then completes the
transfer or hangs up.
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Central Configuration file: Specify whether to allow a transfer during the proceeding state of a
(provisioning sip.cfg consultation call.
Specify whether a transfer is blind or not.
on page A-80.
Local / Centralized Conferencing

The phone can conference together the local user with the remote parties of a
configurable number of independent calls by using the phone’s local audio
processing resources for the audio bridging. There is no dependency on
network signaling for local conferences.
All phones support three-party local conferencing. The SoundPoint IP 450,
550, 560, 650, and 670 and SoundStation IP 7000 phones may support four-way
local conferencing.
Note Four-party conferencing requires a license key for activation. For more information,
refer to Manage Conferences on page 4-22.
If the initiator of a three-party local conference ends the call, the other members of
the call may still communicate. If the initiator of a four-party local conference ends
the call, the conference ends.
The phone also supports centralized conferences for which external resources
are used such as a conference bridge. This relies on network signaling.
Central Configuration file: Specify the conference hold behavior (all parties on hold or only host
(provisioning sip.cfg is on hold).
server) • For more information, refer to Call Handling Configuration <call/>
on page A-80.
Specify whether or not all parties hear sound effects while setting up a
conference.
on page A-80.
Specify which type of conference to establish and the address of the
centralized conference resource.
• For more information, refer to Conference Setup <conference/>
on page A-20.
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Manage Conferences
Note This feature is supported on the SoundPoint IP 450, 550, 560, 650, and 670
desktop phones, the SoundStation IP 7000 conference phone, and the Polycom
VVX business media phone.
This feature requires a license key for activation on all phones except the
SoundStation IP 7000 and the Polycom VVX 1500. Using this feature may require
purchase of a license key or activation by Polycom channels. For more information,
contact your Certified Polycom Reseller.
The individual parties within a conference can be managed. New parties can
be added and information about the conference participants can be viewed
(for example, names, phone numbers, send/receive status or media flow,
receive and transmit codecs, and hold status).

server)
Call Forward
The phone provides a flexible call forwarding feature to forward calls to
another destination. Call forwarding can be applied in the following cases:
• Automatically to all calls
• Calls from a specific caller (extension)
• When the phone is busy
• When Do Not Disturb is active
• After an extended period of alerting

The user can elect to manually forward calls while they are in the alerting state
to a predefined or manually specified destination. The call forwarding feature
works in conjunction with the distinctive incoming call treatment feature
(refer to Distinctive Incoming Call Treatment on page 4-7). The user’s ability
to originate calls is unaffected by all call forwarding options. Each registration
has its own forwarding properties.
Server-based call forwarding is active if the feature is enabled on both the
phone and the server and the phone is registered. If server-based call
forwarding is enabled on any of the phone’s registrations, the other
registrations are not affected. Server-based call forwarding disables local Call
Forward and DND features unless configured otherwise.
Server-based call forwarding will behave the same as per-SIP 2.1 feature with
the following exception:
4 - 22
• If server-based call forwarding is enabled, but inactive, and the user

selects the call forward soft key, the “moving arrow” icon does not appear
on the user’s phone (incoming calls are not forwarded).
Note Server-based and local call forwarding are disabled if Shared Call Appearance or
Bridged Line Appearance is enabled.
The Diversion field with a SIP header is often used by the call server to inform
the phone of a call’s history. For example, when a phone has been set to enable
call forwarding, the Diversion header allows the receiving phone to indicate
who the call was from, and from which phone number it was forwarded. (For
more information, refer to Header Support on page B-4.) .
or locally:
Central Configuration file: Enable or disable server-based call forwarding.

(provisioning sip.cfg • For more information, refer to SIP <SIP/> on page A-11.
server) Enable or disable local call forwarding behavior when server-based
enabled.
Enable or disable display of Diversion header and the order in which
to display the caller ID and number.
Configuration file: Enable or disable server-based call forwarding as a per-registration

phone1.cfg feature.
• For more information, refer to Registration <reg/>on page A-134.
Set all call diversion settings including a global forward-to contact and
individual settings for call forward all, call forward busy, call forward
no-answer, and call forward do-not-disturb.
• For more information, refer to Diversion <divert/> on page A-142.
Local Web Server Set all call diversion settings.

(if enabled) Navigate to: http://<phoneIPAddress>/reg.htm
Local Phone User The user can set the call-forward-all setting from the idle display
Interface (enable/disable and specify the forward-to contact) as well as divert
callers while the call is alerting.
4 - 23
Directed Call Pick-Up

Calls to another phone can be picked up by dialing the extension of the other
phone. This feature depends on support from a SIP server. With many SIP
servers, directed call pick-up is implemented using a particular star code
sequence. With some SIP servers, specific network signaling is used to
implement this feature.

server) Determine the type of directed call pickup.
on page A-80.
Determine the type of SIP header to include.
• For more information, refer to Protocol <voIpProt/> on page A-7.
Group Call Pick-Up

Calls to another phone within a pre-defined group can be picked up without
dialing the extension of the other phone. This feature depends on support from
a SIP server. With many SIP servers, group call pick-up is implemented using
a particular star code sequence. With some SIP servers, specific network
signaling is used to implement this feature.

server)
Call Park/Retrieve
An active call can be parked, and the parked call can be retrieved by another
phone. This feature depends on support from a SIP server. With many SIP
servers, this feature is implemented using a particular star code sequence.
With some SIP servers, specific network signaling is used to implement this
feature.

server) Determine the type of call park and retrieval string.
on page A-80.
4 - 24
Last Call Return

The phone allows server-based last call return. This feature depends on
support from a SIP server. With many SIP servers, this feature is implemented
using a particular star code sequence. With some SIP servers, specific network
signaling is used to implement this feature.

server) Specify the string sent to the server for last-call-return.
on page A-80.
Setting Up Advanced Features

following advanced features:
• Configurable Feature Keys
• Multiple Line Keys per Registration
• Multiple Call Appearances
• Customizable Fonts and Indicators
• Instant Messaging
• Multilingual User Interface
• Downloadable Fonts
• Synthesized Call Progress Tones
• Browser and Microbrowser
• Real-Time Transport Protocol Ports
• Network Address Translation
• Corporate Directory
• Recording and Playback of Audio Calls
• Digital Picture Frame
• Enhanced Feature Keys
• Configurable Soft Keys
• LCD Power Saving
4 - 25
This section also provides information for making configuration changes for
the following advanced call server features:
• Shared Call Appearances
• Bridged Line Appearance
• Busy Lamp Field
• Voice Mail Integration
• Multiple Registrations
• SIP-B Automatic Call Distribution
• Feature Synchronized Automatic Call Distribution
• Server Redundancy
• Presence
• Microsoft Live Communications Server 2005 Integration
• Access URL in SIP Message
• Static DNS Cache
• Display of Warnings from SIP Headers
• Quick Setup of SoundPoint IP / SoundStation IP / VVX Phones
Configurable Feature Keys

All key functions can be changed from the factory defaults. The scrolling
timeout for specific keys can be configured.
Note Since there is no Redial key on the SoundPoint IP 32x/33x phone, the redial
function cannot be remapped.
The rules for remapping of key functions are:
• The phone keys that have removable key caps can be mapped to the
following:
— Any function that is implemented as a removable key cap on any of
the phones (Directories, Applications, Conference, Transfer, Redial,
Menu, Messages, Do Not Disturb, Call Lists)
— A speed-dial
— An enhanced feature key operation
— Null
4 - 26
• The phone keys without removable key caps cannot be remapped. These
include:
— Any keys on the dial pad
— Volume control
— Handsfree, Mute, Headset
— Hold
— Navigation Cluster
Central Configuration File: Set the key scrolling timeout, key functions, and sub-pointers for each
(provisioning sip.cfg key (usually not necessary).
server) • For more information, refer to Keys <key/> on page A-94.
For more information on the default feature key layouts, refer to Default
Feature Key Layouts on page C-13.
Multiple Line Keys per Registration

More than one Line Key can be allocated to a single registration (phone
number or line) on SoundPoint IP and Polycom VVX 1500 phones. The
number of Line Keys allocated per registration is configurable.
or locally:
Central Configuration file: Specify the number of line keys to assign per registration.
(provisioning phone1.cfg • For more information, refer to Registration <reg/> on page A-134.
server)
Local Web Server Specify the number of line keys to assign per registration.
(if enabled) Navigate to http://<phoneIPAddress>/reg.htm
Local Phone User Specify the number of line keys to assign per registration using the
Interface Line Configuration menu. Either the Web Server or the provisioning
server configuration files or the local phone user interface should be
used to configure registrations, not a mixture of these options. When
the Line Configuration menu is used, it is assumed that all
registrations use the same server.
4 - 27
Multiple Call Appearances

The phone supports multiple concurrent calls. The hold feature can be used to
pause activity on one call and switch to another call. The number of concurrent
calls per line key is configurable. Each registration can have more than one line
key assigned to it (refer to the previous section, Multiple Line Keys per
Registration).
or locally:
Central Configuration file: Specify the default number of calls that can be active or on hold per
(provisioning sip.cfg line key.
server) • For more information, refer to Call Handling Configuration <call/>
on page A-80.
Configuration file: Specify per-registration the number of calls that can be active or on
phone1.cfg hold per line key assigned to that registration. This will override the
default value specified in sip.cfg.
Local Web Server Specify the default number of calls that can be active or on hold per
(if enabled) line key and the number of calls per registration that can be active or
on hold per line key assigned to that registration.
Navigate to http://<phoneIPAddress>/appConf.htm#ls and
http://<phoneIPAddress>/reg.htm
Local Phone User Specify per-registration the number of calls that can be active or on
Interface hold per line key assigned to that registration using the Line
Configuration menu. Either the Web Server or the provisioning
Customizable Fonts and Indicators

The phone’s user interface can be customized by changing the fonts and
graphic icons used on the display and the LED indicator patterns. Pre-existing
fonts embedded in the software can be overwritten or new fonts can be
downloaded. The bitmaps and bitmap animations used for graphic icons on
the display can be changed and repositioned. LED flashing sequences and
colors can be changed.
Note Customizable fonts and indicators are not supported on the Polycom VVX 1500.
4 - 28
Central Configuration File: Specify fonts to overwrite existing ones or specify new fonts.
(provisioning sip.cfg • For more information, refer to Fonts <font/> on page A-91.
server)
Specify which bitmaps to use.
• For more information, refer to Bitmaps <bitmap/>on page A-100.
Specify how to create animations and LED indicator patterns.
• For more information, refer to Indicators <ind/> on page A-100.
Instant Messaging
The phone supports sending and receiving instant text messages. The user is
alerted to incoming messages visually and audibly. The user can view the
messages immediately or when it is convenient. For sending messages, the
user can either select a message from a preset list of short messages or an
alphanumeric text entry mode allows the typing of custom messages using the
dial pad. Message sending can be initiated by replying to an incoming
message or by initiating a new dialog. The destination for new dialog
messages can be entered manually or selected from the contact directory, the
preferred method.

server)
Multilingual User Interface

The system administrator or the user can select the language. Support for
major western European languages is included and additional languages can
be easily added. Support for Asian languages (Chinese, Japanese, and Korean)
is also included, but will display only on the higher resolution displays of the
SoundPoint IP 450, 550, 560, 650, and 670, SoundStation IP 6000, and 7000, and
Polycom VVX 1500. A WGL4 character set is displayed the SoundStation IP
7000. For more information, refer to
http://www.microsoft.com/OpenType/otspec/WGL4E.HTM.
For basic character support and extended character support (available on
SoundPoint IP 450, 550, 560, 650 and 670 and SoundStation IP platforms), refer
to Multilingual <ml/> on page A-28. (Note that within a Unicode range, some
characters may not be supported due to their infrequent usage.)
The SoundPoint IP and SoundStation IP user interface is available in the
following languages by default: Simplified Chinese (if displayable), Danish,
Dutch, English, French, German, Italian, Japanese (if displayable), Korean (if
displayable), Norwegian, Polish, Brazilian Portuguese, Russian, Slovenian,
International Spanish, and Swedish.
4 - 29
Note The multilingual feature relies on dictionary files resident on the provisioning server.
The dictionary files are downloaded from the provisioning server whenever the
language is changed or at boot time when a language other than the internal US
English language has been configured. If the dictionary files are inaccessible, the
language will revert to the internal language.
Note Currently, the multilingual feature is only available in the SIP application. The
BootROM application is available in English only.

or locally:
Central Configuration file: Specify the boot-up language and the selection of language choices
(provisioning sip.cfg to be made available to the user.
server) • For more information, refer to Multilingual <ml/> on page A-28.
For instructions on adding new languages, refer to To add new
languages to those included with the distribution: on page A-29.
Local Local Phone User The user can select the preferred language under the Settings menu.
Interface The languages appears in the list in the language itself. For example,
German appears in the list as “Deutsch” and Swedish appears as
“Svenska”. For administrator convenience, the ISO representation of
each language is also included in the language selection menu.
Downloadable Fonts
New fonts can be loaded onto the phone. For guidelines on downloading
fonts, refer to Fonts <font/> on page A-91.
Note Downloadable fonts are not supported on the SoundStation IP 6000 and 7000 and
the Polycom VVX 1500.
Synthesized Call Progress Tones

In order to emulate the familiar and efficient audible call progress feedback
generated by the PSTN and traditional PBX equipment, call progress tones are
synthesized during the life cycle of a call. These call progress tones are easily
configurable for compatibility with worldwide telephony standards or local
preferences.
4 - 30
Central Configuration file: Specify the basic tone frequencies, levels, and basic repetitive
(provisioning sip.cfg cadences.
server) • For more information, refer to Chord-Sets <chord/> on page A-36.
Specify downloaded sampled audio files for advanced call progress
tones.
• For more information, refer to Sampled Audio for Sound Effects
<saf/> on page A-37.
Specify patterns.
• For more information, refer to Patterns <pat/> on page A-39 and
Call Progress Patterns on page A-40.
Browser and Microbrowser

The SoundPoint IP 430, 450, 550, 560, 650, and 670 phones, the SoundStation IP
6000, and 7000 phones, and the Polycom VVX 1500 phones (running releases
before SIP 3.2.2) support an XHTML Microbrowser. This can be launched by
pressing the Applications key or it can be accessed through the Menu key by
selecting Applications.
Note On some older phones, the Applications key is labeled Services.
The Polycom VVX 1500 phones running SIP 3.2.2 or later support a full
browser. This can be launched by pressing the App key or it can accessed
through the Menu key by selecting Applications.
Note If Browser uses over 30MB of memory and either the amount of free memory on the
phone is below 6MB or the real time is between 1am to 5am, then the browser will
restart. Once the browser has restarted, the last displayed web page is restored.
Two instances of the Microbrowser or Browser may run concurrently:
• An instance with standard interactive user interface
• An instance that does not support user input, but appears in a window on
the idle display
For more information, refer to the Web Application Developer’s Guide, which can
be found at http://www.polycom.com/voicedocumentation/.
4 - 31

or locally:
Central Configuration file: Specify the Application browser home page, a proxy to use, and size
(provisioning sip.cfg limits.
server) • For more information, refer to Microbrowser <mb/> on page
A-119.
Specify the telephone notification and state polling events to be
recorded and location of the push server.
• For more information, refer to Applications <apps/> on page
A-122.
Local Web Server Specify the Applications browser home page and proxy to use.
(if enabled) Navigate to http://<phoneIPAddress>/coreConf.htm#mb
Real-Time Transport Protocol Ports

The phone is compatible with RFC 1889 - RTP: A Transport Protocol for
Real-Time Applications - and the updated RFCs 3550 and 3551. Consistent
with RFC 1889, the phone treats all RTP streams as bi-directional from a
control perspective and expects that both RTP end points will negotiate the
respective destination IP addresses and ports. This allows real-time transport
control protocol (RTCP) to operate correctly even with RTP media flowing in
only a single direction, or not at all. It also allows greater security: packets from
unauthorized sources can be rejected.
The phone can filter incoming RTP packets arriving on a particular port by IP
address. Packets arriving from a non-negotiated IP address can be discarded.
The phone can also enforce symmetric port operation for RTP packets: packets
arriving with the source port set to other than the negotiated remote sink port
can be rejected.
The phone can also fix the destination transport port to a specified value
regardless of the negotiated port. This can be useful for communicating
through firewalls. When this is enabled, all RTP traffic will be sent to the
specified port and will be expected to arrive on that port as well. Incoming
packets are sorted by the source IP address and port, allowing multiple RTP
streams to be multiplexed.
The RTP port range used by the phone can be specified. Since conferencing
and multiple RTP streams are supported, several ports can be used
concurrently. Consistent with RFC 1889, the next higher odd port is used to
send and receive RTCP.
4 - 32

or locally:
Central Configuration file: Specify whether to filter incoming RTP packets by IP address,
(provisioning sip.cfg whether to require symmetric port usage or whether to jam the
server) destination port and specify the local RTP port range start.
• For more information, refer to RTP <rtp/> on page A-73.
Local Web Server Specify whether to filter incoming RTP packets by IP address,
(if enabled) whether to require symmetric port usage, whether to jam the
destination port and specify the local RTP port range start.
Navigate to: http://<phoneIPAddress>/netConf.htm#rt
Network Address Translation

The phone can work with certain types of network address translation (NAT).
The phone’s signaling and RTP traffic use symmetric ports (the source port in
transmitted packets is the same as the associated listening port used to receive
packets) and the external IP address and ports used by the NAT on the phone’s
behalf can be configured on a per-phone basis.
or locally:
Central Configuration file: Specify the external NAT IP address and the ports to be used for
(provisioning sip.cfg signaling and RTP traffic.
server) • For more information, refer to Network Address Translation
<nat/> on page A-150.
Local Web Server Specify the external NAT IP address and the ports to be used for
(if enabled) signaling and the RTP traffic.
Navigate to: http://<phoneIPAddress>/netConf.htm#na
4 - 33
Corporate Directory
Note This feature requires a license key for activation except on the SoundStation IP
7000 and Polycom VVX 1500. Using this feature may require purchase of a license
key or activation by Polycom channels. For more information, contact your Certified
Polycom Reseller.
The SoundPoint IP, SoundStation IP, and Polycom VVX phones can be
configured to interface with a corporate directory server that supports the
Lightweight Directory Access Protocol (LDAP) version 3. Currently the
following LDAP servers are supported:
• Microsoft Active Directory 2003
• Sun ONE Directory Server 5.2 p6
• Open LDAP Directory Server 2.4.12
• Microsoft Active Directory Application Mode (ADAM) 1.0 SP1

Both corporate directories that support server-side sorting and those that do
not are supported. In the latter case, the sorting is performed on the phone.
Polycom recommends using corporate directories that have server-side sorting.

Polycom recommends that you consult your LDAP Administrator when making any
configuration changes for this feature.
The corporate directory can be browsed or searched. Entries retrieved from the
LDAP server can be saved to the local contact directory on the phone. Phone
calls can be placed based on the phone number contained in the LDAP entry.
The corporate directory interface is read only, so that editing or deleting
existing directory entries as well as adding new directory entries from the
phone is not be possible. (There is no matching of first and last names in the
corporate directory to incoming calls, caller identification display, and in the
call lists.)
All attributes are considered to be Unicode text. Validity checking will be
performed when a call is placed or the entry is saved to the local contact
directory.
The corporate directory LDAP server status can be reviewed through the
Status menu (Status > CD Server Status).
For detailed examples for all currently supported LDAP directories, refer to
“Technical Bulletin 41137: Best Practices When Using Corporate Directory on
SoundPoint IP / SoundStation IP / VVX Phones“ at
4 - 34

or locally:
Central Configuration file: Specify the location of the corporate directory’s LDAP server, the
(provisioning sip.cfg LDAP attributes, how often to refresh the local cache from the LDAP
server) server, and other miscellaneous parameters.
• For more information, refer to Corporate Directory <corp/> on
page A-87.
Local Local Phone User Enable or disable persistent viewing through the Settings menu
Interface (Settings > Basic > Preferences > Corporate Directory > View
Persistency).
This section contains the following information:
• Corporate Directory LDAP Attributes
• Browsing the Corporate Directory
Corporate Directory LDAP Attributes

The entry attributes in the corporate directory are mapped through sip.cfg
configuration file attributes to the LDAP attributes first_name, last_name,
phone_number, and others so the SIP application knows how to use them for
searching, dialing, or saving to the local contact directory. Multiple attributes
of the same type are allowed.
Note The maximum of eight attributes can be configured in sip.cfg .
The configuration order dictates how the attributes are displayed and sorted.
The first attribute is the primary sort index and the second attribute is the
secondary sort index. The other attributes are not used in sorting.
To limit the amount of data displayed in the corporate directory, filtering of
the entries can be configured for all attribute types. Filtering can be configured
to be retained if the phone reboots.
For more information on LDAP attributes, refer to RFC 4510 - Lightweight
Directory Access Protocol (LDAP): Technical Specification Road Map.
Browsing the Corporate Directory

The SoundPoint IP or SoundStation IP phone will establish a session with the
corporate directory and download enough entries to fill its cache:
• when the corporate directory is first accessed
4 - 35
• when the phone boots up if the background synchronization parameter is

enabled
The requested entries are based on the configured attributes (see previous
section).
If the background synchronization parameter is enabled, a timer is initiated to
permit a periodic download from the corporate directory.
Entries are sorted according to the order in which the first two attributes are
configured (for example, last name, then first name).
The browse position within the corporate directory as well as the attribute
filters are maintained for subsequent corporate directory access can be saved
(if so configured).
Recording and Playback of Audio Calls
Note This feature requires a license key for activation except for the Polycom VVX 1500.
Using this feature may require purchase of a license key or activation by Polycom
channels. For more information, contact your Certified Polycom Reseller.
The SoundPoint IP 650 and 670 and the Polycom VVX 1500 phones can be
configured to allow recording of audio calls on a supported USB device.
The filenames of the recorded .wav files will include a date/time stamp (for
example, 20Apr2007_190012.wav was created on April 20, 2007 at 19:00:12).
An indication of the recording time remaining—the space available of the
attached USB storage media—appears on the graphic display. The user can
browse through all recorded files through the menu shown on the graphic
display.
Note Notify your users that they may be required by federal, state, and/or local laws to
notify some or all called parties when they are recording.
Playback of recorded files can occur on the phone as well as on other devices,
such as a Windows® or Apple® based computer using an application like
Windows Media Player® or iTunes®.
The user controls which calls are recorded and played back.
For a list of supported USB devices, refer to “Technical Bulletin 38084:
Supported USB Devices for SoundPoint IP 650 and 670 and Polycom VVX 1500
Phones“ at
4 - 36

server)
Digital Picture Frame
Note This feature is only supported on the Polycom VVX 1500.
A slide show of multiple personal images stored on a USB flash drive can be
displayed on the Polycom VVX 1500 phone during the idle mode. The
supported formats include JPEG, BMP, and PNG. The maximum image size is
9999x9999. A maximum of 1000 images can be displayed and these must be
stored in a directory of the USB flash drive that you create.
Note Although 9999x9999 images and progressive/multiscan JPEG images are

supported, the maximum image size that can be downloaded is restricted by the
available memory in the phone.
Central Configuration file: Turn this feature on or off and configure how it appears.
(provisioning sip.cfg • For more information, refer to Feature <feature/> on page A-116
server) • For more information, refer to User Preferences <up/> on page
A-155.
Configuration file: Configure how the feature appears.

phone1.cfg • For more information, refer to User Preferences <up/> on page
A-155.
Note The digital picture frame can be accessed through the PicFrame:// URL.
4 - 37
Enhanced Feature Keys
Note The Enhanced Feature Key feature from SIP 3.0 is compatible with Enhanced
Feature Key feature from SIP 3.0 . However, improvements have been made, and
Polycom recommends that existing configuration files be reviewed and updated.
Customers replacing legacy telephony PBX or key system would like to get
equivalent functionality from their new VoIP telephony system. The enhanced
feature key capability is designed to allow system administrators to program
the speed-dials and soft keys on their phones to interact with the phone user
to implement commonly used functions such as “Call Park” in an intuitive
fashion.
This capability applies to the SoundPoint IP 32x/33x, 430, 450, 550, 560, 650,
and 670 desktop phones and Polycom VVX 1500 business media phones. The
enhanced feature key functionality is implemented using Star Code sequences
and SIP messaging.
The enhanced feature key macro language was designed to follow current
configuration file standards and to be extensible. It is described in more detail
in Enhanced Feature Key Definition Language.
The particular Star Code sequence and the associated prompts displayed on
the SoundPoint IP phone for the enhanced feature are defined by macros.
These macros are case sensitive.
The enhanced feature key capability can be used to provide a customized,
interactive user interface by mapping functions from speed-dial keys, soft
keys and re-mapped hard function keys.
This section provides detailed information on:
• Enhanced Feature Key Definition Language
• Macro Definition
• Configuration File Changes
• Useful Tips
• Examples
For more examples including sample configuration files, refer to “Technical
Bulletin 42250: Using Enhanced Feature Keys and Configurable Soft Keys on
SoundPoint IP, SoundStation IP, and Polycom VVX 1500 Phones” at
4 - 38
Enhanced Feature Key Definition Language

This section defines the additional fields to be entered into a configuration file
for controlling the enhanced feature key behavior. The definition language
follows the XML style notation. The following elements are part of the
definition language:
• <efk/>
• <efklist/>
• <efkprompt/>
• <version/>
• Special Characters
<efk/>
This element indicates the start of enhanced feature key definition section. The
efk element has the following format:
<efk> ... </efk>
<efklist/>
This element describes behavior of enhanced feature key.
The different blocks of the enhanced feature key definitions are uniquely
identified by number following efk.efklist prefix (for example,
efk.efklist.1.<suffix>).
Note In SIP 3.0, a maximum of 50 element groups is supported, however, the exact
number is dependent on available RAM and processing speed. The disabled
elements are included in the total count.
This element contains the following parameters:
Name Interpretation
mname This is the unique identifier that is used for the

speed-dial configuration to reference the enhanced
feature key entry. It cannot start with a digit.
This parameter must have a value and it cannot be Null.
status This parameter has the following values:

• If set to 1, this key is enabled.
• If set to 0 or Null, this key is disabled.
If this parameter is omitted, the value 0 is used.
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Name Interpretation
label This field defines the text string that will be used as a
label on any user text entry screens during enhanced
feature key operation. The value can be any string
including the null string (in this case, no label appears).
If this parameter is omitted, the Null string is used.
Note: If you exceed the phone physical layout text
limits, the text will be shortened and "..." will be
appended.
type The SIP method to be performed once the macro starts

executing. This parameter has the following values:
• If set to “invite “, the action required is performed
using the SIP INVITE method.
Note: This parameter is included for backwards
compatability only. Do not use if at all possible. If the
action.string contains types, this parameter is ignored. If
this parameter is omitted, the default is INVITE.
action.string The action string contains a macro definition of the

action to be performed.
For more information, refer to Macro Definition on page
4-42.
<efkprompt/>
This element describes the behavior of the user prompts.
The different blocks are uniquely identified by number following
efk.efkprompt prefix (for example, efk.efkprompt.1.<suffix>).
Note In SIP 3.0, a maximum of four user prompts were supported. In SIP 3.0, a
maximum of ten user prompts are supported.
This element contains the following parameters:
Name Interpretation
status This parameter has the following values:

• If set to 1, this key is enabled.
• If set to 0, this key is disabled.
Note: If a macro attempts to use a prompt that is
disabled or invalid, the macro execution fails.
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Name Interpretation
label This parameter sets the prompt text that will be

presented to the user on the user prompt screen. The
value can be any string including the null string (in this
case, no label appears).
If this parameter is omitted, the Null string is used.
Note: If you exceed the phone physical layout text
limits, the text will be shortened and "..." will be
appended.
userfeedback This parameter specifies the user input feedback

method. It has the following values:
• If set to “visible”, the text appears as clear text.
• If set to “masked”, the text appears as “*”
characters. For example, if a password is entered.
If this parameter is omitted, the value “visible” is used.
If this parameter has an invalid value (including Null),
this prompt is invalid and all parameters depending on
this prompt are invalid.
type The type of characters entered by the user. This

parameter has the following values:
• If set to “numeric “, the characters are interpreted as
numbers.
• If set to “text”, the characters are interpreted as
letters.
If this parameter is omitted, the value “numeric” is used.
If this parameter has an invalid value (including Null),
this prompt is invalid and all parameters depending on
this prompt are invalid.
Note: A mix of numeric and text is not supported.
<version/>
This element contains the version of the enhanced feature key elements. The
version element has the following format:
<version efk.version=”2”/>
If this parameter is omitted or has an invalid value (including Null), the

enhanced feature key is disabled. This parameter is not required if there are no
efk.efklist entries.
Note In SIP 3.0, “1” is the only supported version. In SIP 3.1 or later, “2” is the only
supported version.
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Special Characters
The following special characters are used to implement the enhanced feature
key functionality:
• ! — The characters following it are a macro name.
• ' or ASCII (0x27) — This character delimits the commands within the
macro.
• $ — This character delimits the parts of the macro string. This character
must exist in pairs, where the delimits the characters to be expanded.
• ^ — This character indicates that the following characters represent the

expanded macro (as in the action string).
Macro names and action strings cannot contain these characters. If they do,
unpredictable results may occur.
Macro Definition
The action.string in the efklist element can be defined by either:
• Macro Action
• Prompt Macro Substitution
• Expanded Macros
Macro Action
The action string is executed in the order it appears. User input is collected
before any action is taken.
The action string contains the following fields:
Name Interpretation
$L<label>$ This is the label for the entire operation. The value can
be any string including the null string (in this case, no
label appears). This label will be used if no other
operation label collection method worked (up to the
point where this field is introduced). Make this the first
entry in action string to be sure this label is used;
otherwise another label may be used and this one
ignored.
digits The digits to be sent.

The appearance of this this parameter depends on the
action string.
4 - 42
Name Interpretation
$C<command>$ This is the command. It can appear anywhere in the

action string.
Supported commands (or shortcuts) include:
• hangup (hu)
• hold (h)
• waitconnect (wc)
• pause <number of seconds> (p <num sec>) where
the maximum value is 10
$T<type>$ The embedded action type. Multiple actions can be

defined.
Supported action types include:
• invite
• dtmf
• refer
Note: Polycom recommends that you always define this
field. If it is not defined, the supplied digits will be dialed
using INVITE (if no active call) or DTMF (if an active
call). The use of refer method is call server dependent
and may require the addition of star codes.
$M<macro>$ The embedded macro. The <macro> string must begin

with a letter.
If the macro name is not defined, the execution of the
action string fails.
$P<prompt num>N<num The user input prompt string.

digits>$ Refer to Prompt Macro Substitution on this page.
$S<speed dial index>$ The speed dial index. Only digits are valid.
The action is found in the contact field of the local
directory entry pointed to by the index.
$F<internal function>$ An internal function.

For more information, refer to Internal Key Functions on
page C-19.
URL A URL. Only one per action string is supported.
Prompt Macro Substitution

The action.string in the efklist element can be defined by a macro
substitution string, “PnNn” where:
• Pn is the prompt x as defined in the efk.efkprompt.x
• Nn is the number of digits or letters that the user can enter. The maximum
number is 32. The user needs to press the Enter soft key to complete data
entry.
4 - 43
Note If the maximum number of characters is greater than 32 or less than one, macro
execution fails.
The macros provide a generic and easy to manage way to define the prompt to
be displayed to the user, the maximum number of characters that the user can
input, and action that the phone performs once all user input has been
collected. The macros are case sensitive.
If a macro attempts to use a prompt that is disabled, the macro execution fails.
A prompt is not required for every macro.
Expanded Macros
Expanded macros are prefixed with the “^” character and are inserted directly
into the local directory contact field. For more information, refer to Local
Contact Directory File Format on page 4-11.
Configuration File Changes
Note The configuration file changes and the enhanced feature key definitions can be
included together in one configuration file.
A sample configuration for this feature—including the enhanced feature keys
definitions shown in the following section, Examples— may be included with the
SIP 3.1 release.
Create a new configuration file in the style of sip.cfg in order to make configuration
changes. For more information on why to create another configuration file,refer to
the “Configuration File Management on SoundPoint IP, SoundStation IP, and
Polycom VVX 1500 Phones” white paper at
Configuration changes can be performed centrally at the boot server:

(boot server) sip.cfg • For more information, refer to Feature <feature/> on page A-116.
Configuration file: Specify two calls per line key.

XML file: <Ethernet This file holds the macro names which correspond to the mname fields
address>-directory. in the configuration file where the enhanced feature keys are defined.
xml Macro names must be embedded into the contact (cn) fields with the
“!” prefix. You can also add labels in the first name (fn) fields.
For information on file format, refer to Local Contact Directory File
Format on page 4-11.
4 - 44
Useful Tips
The following information should be noted:
• Activation of the enhanced feature key will fail if configured values are
invalid except where noted in previous sections.
• All failures are logged at level 4 (minor).
• If two macros have the same name, the first one will be used and the
subsequent ones will be ignored.
• “'!” and “^” macro prefixes cannot be mixed in the same macro line.
• A sequence of characters prefixed with “!” are parsed as a macro name.

The exception is the speed dial reference, which starts with “!” and
contains digits only.
• A sequence of characters prefixed with “^” is the action string.
• The sequence of characters accessed from speed dial keys must be prefixed
by either “!” or “^” so it will be processed as an enhanced feature key. All
macro references and action strings added to the local directory contact
field must be prefixed by either “!” or “^”.
• Action strings used in soft key definitions do not need to be prefixed by

“^”. However, the “!” prefix must be used if macros or speed dials are
referenced.
For more information, refer to Configurable Soft Keys on page 4-48.
• A sequence of macro names in the same macro is supported (for example,

“!m1!m2” ).
• A sequence of speed dial references is supported (for example, “!1!2” ).
• A sequence of macro names and speed dial references is supported (for

example, “!m1!2!m2” ).
• Macro names that appear in the local contact directory must follow the
format “!<macro name>” , where <macro name> must match an
<elklist> mname entry. The maximum macro length is 100 characters.
• A sequence of macros is supported, but cannot be mixed with other action

types.
• Action strings that appear in the local contact directory must follow the
format “^<action string>”. Action strings can reference other macros or
speed dial indexes. Protection against recursive macro calls exists (the
enhanced feature keys fails once 50 macro substitutions is reached).
4 - 45
Examples
You must make the the following changes to the <feature/> parameter that
is defined in the sip.cfg configuration file:
<feature feature.18.name="enhanced-feature-keys"
feature.18.enabled="1"/>
Action String Example

The action string
“$Changup$*444*$P1N4$$Tinvite$$Cwaitconnect$$P2N3$$Cpause2$$Tdt
mf$$Changup$” is executed as follows:
1. The user is prompted for 4 digits. For example, “1234”.
2. The user is prompted for 3 digits. For example, “567”.
3. The user’s active call is disconnected.
4. The string “*444*1234” is sent using the INVITE method.
5. Once connected, there is a 2 second pause, and then the string “567” is
sent using DTMF dialing on the active call.
6. The active call is disconnected.
Speed Dial Example
Your organization voice mail system is accessible through 7700 and your voice
mail password is 2154. You could use a speed dial key to access your voice
mail if you entered “7700$Cpause3$2154” as the contact number.
Enhanced Feature Key XML Files
You must ensure that the following XML code exists for the definition of “Call
Park”:
...
<efklist
...
efk.efklist.2.mname="callpark"
efk.efklist.2.status="1"
efk.efklist.2.label="Call Park"
efk.efklist.2.use.idle="1"
efk.efklist.2.use.active="1"
efk.efklist.2.use.alerting="1"
efk.efklist.2.use.dialtone="1"
efk.efklist.2.use.proceeding="1"
efk.efklist.2.use.setup="1"
efk.efklist.2.type="invite"
efk.efklist.2.action.string="*68*$P1N10$"
...
/>
<efkprompt
efk.efkprompt.1.status="1"
4 - 46
efk.efkprompt.1.label="Enter Number: "

efk.efkprompt.1.userfeedback="visible"
efk.efkprompt.1.type="numeric"
efk.efkprompt.1.digitmatching="style1"
...
/>
...
Contact Directory Changes

You must make the following contact directory changes for the definition of
“Call Park”:
<directory>
<item_list>
<item>
<fn>Call Park</fn>
<ct>!callpark</ct>
<sd>2</sd>
<rt>4</rt>
<ad>0</ad>
<ar>0</ar>
<bw>0</bw>
<bb>0</bb>
</item>
</item_list>
</directory>
Note To avoid users accidently deleting the definitions in the contact directory, make the
contact directory read only. For more information, refer to Local Directory <local/>
on page A-86.
Using Call Park Key

The following figure shows the second speed dial key mapped to Call Park (as
well as others mapped to Park Return and Call Pickup).
4 - 47
To use the Call Park key during an active call:

1. When there is an active call on line 2233:
a Select the Call Park soft key.
The Call Park screen appears.
b Enter the number where you want to park the active call, then select
the Enter soft key.
The Call Park * code (*68) is prepended to the number you entered and
the call is parked at that location by the call server. The active call is
put on hold during this operation.
Configurable Soft Keys

This feature enables phone system administrators to “program” certain
frequently used functions onto the soft keys at the bottom of the phone
display. This programming can be controlled based on call state. For example
a Call Park function can be presented to the user when in an active call state.
If certain hard keys are missing, you may want to create a soft key. For
example, if there is no Do Not Disturb key on a phone, you could create a Do
Not Disturb soft key.
New soft keys can be mapped into:
• An Enhanced Feature Key sequence
• A speed dial contact directory entry
4 - 48
• Directly into the Enhanced Feature Key macro
• Directly into a URL
• A chained list of actions

It is possible to disable the display of specific standard keys—the soft keys that
are displayed on SoundStation IP, SoundStation IP, and Polycom VVX 1500
phones—to make room for other soft keys that your organization wants
displayed. To ensure that the usability of features is not compromised, the
disabling of certain soft keys in certain circumstances may be restricted. When
a standard soft key is disabled, the space where it was remains empty. The
standard keys that can be disabled include:
• New Call
• End Call
• Split
• Join
• Forward
• Directories (or Dir as it is called on the SoundPoint IP 32x/33x)
• Callers (appears on the SoundPoint IP 32x/33x)
• MyStatus and Buddies
• Hold, Transfer, and Conference
Note The Hold, Transfer, and Conference are grouped together to avoid usability
issues.
Custom soft keys can be added in the following call states:
• Idle—There are no active calls.
• Active—This state starts when a call is connected. It stops when the call
stops or changes to another state (like hold or dial tone).
• Alerting (or ringing or incoming proceeding)—The phone is ringing.
• Dial tone—You can hear the dial tone.
• Proceeding (or outgoing proceeding)—This state starts when the phone

sends a request to the network. It stops when the call is connected.
• Setup—This state starts when the user starts keying in a phone number.
This state ends when the Proceeding state starts.
• Hold—The call is put on hold locally.
4 - 49
Custom soft keys can be configured to precede the standard soft keys that are
still displayed. The order of the custom soft keys follows the configuration
order. The standard soft keys are shifted to the right and any empty spaces are
removed.
If the custom soft keys are configured to not precede the standard soft keys,
then the standard soft keys do not move. The order of the custom soft keys—
starting from the leftmost empty space—follows the empty spaces. Any extra
custom soft keys that are left after all empty spaces are used are appended at
the end.
Up to 10 soft keys can be configured. Any additional soft keys are ignored. If
more soft keys are defined than fit on the graphic display at one time, a More
soft key is displayed followed by the remainder of the soft keys that you have
defined.
This capability applies to the SoundPoint IP 32x, 33x, 430, 450, 550, 560, 650,
and 670, and Polycom VVX 1500 phones. This capability is linked to the
Enhanced Feature Key feature (refer to Enhanced Feature Keys on page 4-38.)

(boot server) sip.cfg • For more information, refer to Feature <feature/> on page A-116.
Specify the soft key label, in what states it should be displayed, and
prompt for input if required.
• For more information, refer to Soft Keys <softkey/> on page
A-128.
Configuration File Examples

For more examples, refer to “Technical Bulletin 42250: Using Enhanced
Feature Keys and Configurable Soft Keys on SoundPoint IP, SoundStation IP,
and Polycom VVX 1500 Phones” at
To disable the New Call soft key:

1. Update the sip.cfg configuration as follows:
softkey.feature.newcall = 0
2. Reboot the phone.

The New Call soft key is not displayed and the space where it usually
appears is empty.
4 - 50
To map a chained list of actions to a soft key:

1. Configure speed dial index 2 in contact directory with a regular phone
number. For example, enter “2900” in the contact field.
2. Configure speed dial index 1 in contact directory with “!2” in contact
field.
3. Update the sip.cfg configuration as follows:
softkey.1.label = ChainAct
softkey.1.action = $S1$$Tinvite$
softkey.1.use.idle = 1

If you press the soft key ChainAct, the phone dials number 2900.
To map the Do Not Disturb Enhanced Feature Key sequence to a soft key:
1. Update sip.cfg as follows:
softkey.1.label = DND
softkey.1.action = $FDoNotDisturb$
softkey.1.use.idle = 1

A DND soft key is displayed on the phone when it is in the idle state.
When the DND soft key is pressed, the Do Not Disturb icon is displayed.
To map a Send to Voice Mail Enhanced Feature Key sequence to a soft key:
Note The exact star code to transfer the active call to Voice Mail depends on your call
server.
1. Update sip.cfg as follows:
softkey.2.label = ToVMail
softkey.2.action = ^*55$P1N10$$Tinvite$
softkey.2.use.alerting = 1

When another party calls, the ToVMail soft key is displayed. When the
user presses ToVMail soft key, the other party is transferred to voice mail.
4 - 51
LCD Power Saving
Note This feature is only supported on the Polycom VVX 1500.
This feature applies during configured non-working hours and when the
phone is idle. Working hours are defined in the configuration files and users
can change the default values through the phone’s menu to accommodate
their individual schedules. The Polycom VVX 1500 phone enters
power-saving mode after it has been idle for a certain period of time and its
camera doesn’t detect motion. The phone’s ability to detect the users’ presence
is biased for easy detection during office hours and for difficult detection
during off hours.
Central Configuration file: Turn this feature on or off and configure how it works.
(provisioning sip.cfg • For more information, refer to LCD Power Saving
server) <powerSaving/> on page A-132.
Shared Call Appearances

Calls and lines on multiple phones can be logically related to each other. A call
that is active on one phone will be presented visually to phones that share that
call appearance. Mutual exclusion features emulate traditional PBX or key
system privacy for shared calls. Incoming calls can be presented to multiple
phones simultaneously. Users at the different locations have the ability to
interrupt remote active calls.
This feature is dependent on support from a SIP server that binds the
appearances together logically and looks after the necessary state notifications
and performs an access control function. For more information, refer to Shared
Call Appearance Signaling on page B-10.
Note Shared call appearances and bridged line appearances are two different signaling
methods of implementing a feature whereby more than one phone can share the
same line or registration. These implementations are dependent on the SIP server.
The methods are mutually exclusive and you should confirm with the call server
vendor which (if any) method is supported.
4 - 52

or locally:
Central Configuration file: Specify whether diversion should be disabled on shared lines.
(provisioning sip.cfg • For more information, refer to Shared Calls <shared/> on page
server) A-84.
Specify line-seize subscription period.
• For more information, refer to Server <server/> on page A-8.
Specify standard or non-standard behavior for processing line-seize
subscription for mutual exclusion feature.
• For more information, refer to Special Events <specialEvent/> on
page A-20.
Configuration file: Specify per-registration line type (private or shared), barge-in

phone1.cfg capabilities, and line-seize subscription period if using per-registration
servers. A shared line will subscribe to a server providing call state
information.
Specify per-registration whether diversion should be disabled on
shared lines.
Local Web Server Specify line-seize subscription period.

(if enabled) Navigate to http://<phoneIPAddress>/appConf.htm#se
Specify standard or non-standard behavior for processing line-seize
subscription for mutual exclusion feature.
Navigate to http://<phoneIPAddress>/appConf.htm#ls
Specify per-registration line type (private or shared) and line-seize
subscription period if using per-registration servers, and whether
diversion should be disabled on shared lines.
Navigate to http://<phoneIPAddress>/reg.htm
Local Phone User Specify per-registration line type (private or shared) using the Line
Interface Configuration menu. Either the Web Server or the provisioning
4 - 53
Bridged Line Appearance

Calls and lines on multiple phones can be logically related to each other. A call
that is active on one phone will be presented visually to phones that share that
line. Incoming calls can be presented to multiple phones simultaneously. This
feature is dependent on support from a SIP server that binds the appearances
together logically and looks after the necessary state notifications and
performs an access control function. For more information, refer to Bridged
Line Appearance Signaling on page B-10.
Note Bridged line appearances and shared call appearances are two different signaling
methods of implementing a feature whereby more than one phone can share the
same line or registration. These implementations are dependent on the SIP server.
The methods are mutually exclusive and you should confirm with the call server
vendor which (if any) method is supported.
In the configuration files, bridged lines are configured by “shared line” parameters.

or locally:
Central Configuration file: Specify whether diversion should be disabled on shared lines.
(provisioning sip.cfg • For more information, refer to Call Handling Configuration <call/>
server) on page A-80.
Configuration file: Specify per-registration line type (private or shared) and the shared
phone1.cfg line third party name. A shared line will subscribe to a server
providing call state information.
Specify per-registration whether diversion should be disabled on
shared lines.
Local Web Server Specify per-registration line type (private or shared) and third party
(if enabled) name, and whether diversion should be disabled on shared lines.
Local Phone User Specify per-registration line type (private or shared) and the shared
Interface line third party name using the Line Configuration menu. Either the
Web Server or the provisioning server configuration files or the local
phone user interface should be used to configure registrations, not a
mixture of these options. When the Line Configuration menu is used,
it is assumed that all registrations use the same server.
4 - 54
Busy Lamp Field
Note This feature is available only on SoundPoint IP 430, 450, 550, 560, 600, 601, 650,
and 670 phones. Other SoundPoint IP phone models may be monitored, but cannot
be configured to monitor other phones.
Some aspects of this feature are dependent on the SIP server signaling. You
should consult your SIP server partner or Polycom Channel partner for information
as needed.
The Busy Lamp Field (BLF) feature enhances support for a phone-based
attendant console. It allows monitoring the hook status and remote party
information of users through the busy lamp fields and displays on an
attendant console phone.
In the SIP 3.1 release, the BLF feature was updated for the following:
• Visual and audible indication when a remote line is in an alerting state
• Display of the caller ID of calls on remotely monitored lines
• Single button “Directed Call Pickup” on a remote line

In the SIP 3.2 release, the BLF feature is updated for the following:
• Configurable list of remote parties to a maximum of 47 with configurable

line key labels
• The introduction of configurable default key press actions
• The ability to remove spontaneous call appearances from incoming calls

on monitored lines
Note The SIP 3.2 update to the BLF feature is not supported on the SoundPoint IP 430.
For more information, refer to “Quick Tip 37381: Enhanced BLF“at

Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Microsoft
Live Communications Server 2005 Integration on page 4-64.
Note Use this feature with TCPpreferred transport (refer to Server <server/> on page
A-8). You can also use UDP transport on SoundPoint IP 650 and 670 phones.
4 - 55
Central Configuration file: Specify the list SIP URI and index of the registration which will be
(provisioning phone1.cfg used to send a SUBSCRIBE to the list SIP URI specified in
server) attendant.uri.
• For more information, refer to Attendant <attendant/> on page
A-151.
Specify the list of monitored resources.
• For more information, refer to Resource List <resourceList/> on
page A-152 and Behaviors <behaviors/> on page A-153.
Voice Mail Integration

The phone is compatible with voice mail servers. The subscribe contact and
callback mode can be configured per user/registration on the phone. The
phone can be configured with a SIP URL to be called automatically by the
phone when the user elects to retrieve messages. Voice mail access can be
configured to be through a single key press (for example, the Messages key on
the SoundPoint IP 430, 450, 550, 560, 650, and 670, and the MSG key on the
Polycom VVX 1500). A message-waiting signal from a voice mail server
triggers the message-waiting indicator to flash and the call waiting audio tone
is played through the active audio path.
or locally:
Central Configuration file: For one-touch voice mail access, enable the “one-touch voice mail”
(provisioning sip.cfg user preference.
server) • For more information, refer to User Preferences <up/> on page
A-31.
Configuration file: For one-touch voice mail access, bypass instant messages to remove
phone1.cfg the step of selecting between instant messages and voice mail after
pressing the Messages key on the SoundPoint IP 430, 450, 550,
560, 650, and 670 and the MSG key on the Polycom VVX 1500
(Instant messages are still accessible from the Main Menu).
On a per-registration basis, specify a subscribe contact for solicited
NOTIFY applications, a callback mode (self call-back or another
contact), and the contact to call when the user accesses voice mail.
• For more information, refer to Messaging <msg/> on page A-149.
4 - 56
Local Web Server For one-touch voice mail access, enable the “one-touch voice mail”
(if enabled) user preference and bypass instant messages to remove the step of
selecting between instant messages and voice mail after pressing the
Messages key on the SoundPoint IP 430, 450, 550, 560, 650, and
670 and the MSG key on the Polycom VVX 1500 (Instant messages
are still accessible from the Main Menu).
Navigate to http://<phoneIPAddress>/coreConf.htm#us
On a per-registration basis, specify a subscribe contact for solicited
NOTIFY applications, a callback mode (self call-back or another
contact) to call when the user accesses voice mail.
Multiple Registrations
The SoundPoint IP 32x/33x and 430 support a maximum of two registrations,
the SoundPoint IP 450 supports three registrations, the SoundPoint IP 550 and
560 support four, and the SoundPoint IP 650 and 670 and the Polycom VVX
1500 support 6. Up to three SoundPoint IP Expansion Modules can be added
to a single host SoundPoint IP 650 and 670 phone increasing the total number
of registrations to 34. The SoundStation IP 6000 and 7000 supports a single
registration.
Each registration can be mapped to one or more line keys (a line key can be
used for only one registration). The user can select which registration to use for
outgoing calls or which to use when initiating new instant message dialogs.
or locally:
Central Configuration file: Specify the local SIP signaling port and an array of SIP servers to
(provisioning sip.cfg register to. For each server specify the registration period and the
server) signaling failure behavior.
• For more information, refer to Server <server/> on page A-8 and
Server <server/> on page A-8.
Configuration file: For up to maximum number of registrations, specify a display name,

phone1.cfg a SIP address, an optional display label, an authentication user ID
and password, the number of line keys to use, and an optional array
of registration servers. The authentication user ID and password are
optional and for security reasons can be omitted from the
configuration files. The local flash parameters will be used instead.
The optional array of servers and their associated parameters will
override the servers specified in sip.cfg if non-Null.
4 - 57
Local Web Server Specify the local SIP signaling port and an array of SIP servers to
(if enabled) register to.
Navigate to http://<phoneIPAddress>/appConf.htm#se
For up to six registrations (depending on the phone model, in this
case the maximum is six even for the IP 650 and 670), specify a
display name, a SIP address, an optional display label, an
authentication user ID and password, the number of line keys to use,
and an optional array of registration servers. The authentication user
ID and password are optional and for security reasons can be omitted
from the configuration files. The local flash parameters will be used
instead. The optional array of servers will override the servers
specified in sip.cfg in non-Null. This will also override the servers on
the appConf.htm web page.
Local Local Phone User Use the Call Server Configuration and Line Configuration menu to
(continued) Interface specify the local SIP signaling port, a default SIP server to register to
and registration information for up to twelve registrations (depending
on the phone model). These configuration menus contains a sub-set
of all the parameters available in the configuration files.
Either the Web Server or the provisioning server configuration files or
the local phone user interface should be used to configure
registrations, not a mixture of these options. When the Line
Configuration menu is used, it is assumed that all registrations use
the same server.
For more information, refer to Server <server/> on page A-8, Server
<server/> on page A-8, and Registration <reg/> on page A-134.
SIP-B Automatic Call Distribution
Note For more information on SIP-B and supported features on SoundPoint IP,
SoundStation IP, and Polycom VVX phones, contact Polycom Product
Management.
The phone allows Automatic Call Distribution (ACD) login and logout. This
feature depends on support from a SIP server.
4 - 58

server)
Configuration file: Enable this feature per registration.
The phone also supports ACD agent availability. This feature depends on
support from a SIP server.

server)
Configuration file: Enable this feature per registration.
Feature Synchronized Automatic Call Distribution

As of SIP 3.1.2, you can use your SoundPoint IP phones in a call center
agent/supervisor role on a supported call server.
When this feature is enabled, the phone will indicate the ACD Call Center
Agent state as directed by the call server. The call center agent is provided with
an entry method to initiate Sign In/Sign Out and other ACD states through
soft keys, however, the phone state will only change once the server has
acknowledged that the phone can move into that new state—in this way, the
ACD state is maintained in synchronization with the call server and any ACD
computer-based soft-clients. The SIP signaling used for this implementation is
described in the Device Key Synchronization Requirements Document;
Release R14 sp2; Document version 1.6. Contact Polycom Product
Management for more information.
The Feature Synchronized ACD feature is supported on SoundPoint IP
32x/33x, 430, 450, 550, 560, 650, and 670, SoundStation IP 6000 and 7000, and
Polycom VVX 1500 phones.
Note The Feature Synchronized ACD feature is distinct from the existing SIP-B
Automatic Call Distribution functionality, which was added in SIP 1.6 .
For details on how to configure SoundPoint IP, SoundStation IP, and VVX
phones for Feature Synchronized ACD, refer to “Technical Bulletin 34787:
Using Feature Synchronized Automatic Call Distribution with SoundPoint IP
and Polycom VVX 1500 Phones” at
4 - 59
Central Configuration file: Enable or disable Feature Synchronized ACD.

(boot server) sip.cfg • For more information, refer to SIP <SIP/> on page A-11.
Turn this feature on or off.
• For more information, refer to Feature <feature/> on page A-116.
Configuration file: Set the registration to be used for Feature Synchronized ACD and the
phone1.cfg users sign-in state.
• For more information, refer to Automatic Call Distribution
<acd/>on page A-156.
Server Redundancy
Server redundancy is often required in VoIP deployments to ensure continuity
of phone service for events where the call server needs to be taken offline for
maintenance, the server fails, or the connection between the phone and the
server fails.
Two types of redundancy are possible:
• Fail-over: In this mode, the full phone system functionality is preserved by

having a second equivalent capability call server take over from the one
that has gone down/off-line. This mode of operation should be done
using DNS mechanisms or “IP Address Moving” from the primary to the
back-up server.
• Fallback: In this mode, a second less featured call server (router or

gateway device) with SIP capability takes over call control to provide basic
calling capability, but without some of the richer features offered by the
primary call server (for example, shared lines, presence, and Message
Waiting Indicator). Polycom phones support configuration of multiple
servers per SIP registration for this purpose.
In some cases, a combination of the two may be deployed.
Note Your SIP server provider should be consulted for recommended methods of
configuring phones and servers for fail-over configuration.
Warning Prior to SIP 2.1, the reg.x.server.y parameters (refer to Registration <reg/> on
page A-134) could be used for fail-over configuration. The older behavior is no
longer supported. Customers that are using the reg.x.server.y. configuration
parameters where y>=2 should take care to ensure that their current deployments
are not adversely affected. For example the phone will only support advanced SIP
features such as shared lines, missed calls, presence with the primary server (y=1).
4 - 60
For more information, refer to “Technical Bulletin 5844: SIP Server Fallback
Enhancements on SoundPoint IP Phones” at
Central Configuration file: Specify global primary and fallback server configuration parameters.
(provisioning sip.cfg • For more information, refer to Protocol <voIpProt/> on page A-7.
server)
Configuration file: Specify per registration primary and fallback server configuration
phone1.cfg parameters values that override those in sip.cfg.
DNS SIP Server Name Resolution

If a DNS name is given for a proxy/registrar address, the IP address(es)
associated with that name will be discovered as specified in RFC 3263. If a port
is given, the only lookup will be an A record. If no port is given, NAPTR and
SRV records will be tried, before falling back on A records if NAPTR and SRV
records return no results. If no port is given, and none is found through DNS,
5060 will be used.
Refer to http://www.ietf.org/rfc/rfc3263.txt for an example.
Note Failure to resolve a DNS name is treated as signaling failure that will cause a
failover.
Behavior When the Primary Server Connection Fails

For Outgoing Calls (INVITE Fallback)
When the user initiates a call, the phone will go through the following steps to
connect the call:
1. Try to make the call using the working server.
2. If the working server does not respond correctly to the INVITE, then try
and make a call using the next server in the list (even if there is no current
registration with these servers). This could be the case if the Internet
connection has gone down, but the registration to the working server has
not yet expired.
3. If the second server is also unavailable, the phone will try all possible
servers (even those not currently registered) until it either succeeds in
making a call or exhausts the list at which point the call will fail.
At the start of a call, server availability is determined by SIP signaling failure.
SIP signaling failure depends on the SIP protocol being used as described
below:
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• If TCP is used, then the signaling fails if the connection fails or the Send
fails.
• If UDP is used, then the signaling fails if ICMP is detected or if the signal
times out. If the signaling has been attempted through all servers in the list
and this is the last server, then the signaling fails after the complete UDP
timeout defined in RFC 3261. If it is not the last server in the list, the
maximum number of retries using the configurable retry timeout is used.
For more information, refer to Server <server/> on page A-8 and
Registration <reg/> on page A-134.
Warning If DNS is used to resolve the address for Servers, the DNS server is unavailable,
and the TTL for the DNS records has expired, the phone will attempt to contact the
DNS server to resolve the address of all servers in its list before initiating a call.
These attempts will timeout, but the timeout mechanism can cause long delays (for
example, two minutes) before the phone call proceeds “using the working server”.
To mitigate this issue, long TTLs should be used. It is strongly recommended that
an on-site DNS server is deployed as part of the redundancy solution.
Hosted VoIP Service

Provider
Call Server 1B
Call Server 1A
Internet
DNS Server
VoIP SMB Customer

SIP Capable Router
Premise
Server2
`
PSTN
PSTN Gateway
`
Phone Configuration
The phones at the customer site are configured as follows:
• Server 1 (the primary server) will be configured with the address of the
service provider call server. The IP address of the server(s) to be used will
be provided by the DNS server. For example:
reg.1.server.1.address="voipserver.serviceprovider.com"
• Server 2 (the fallback server) will be configured to the address of the

router/gateway that provides the fallback telephony support and is
on-site. For example:
reg.1.server.2.address=172.23.0.1
4 - 62
Note It is possible to configure the phone for more than two servers per registration, but
you need to exercise caution when doing this to ensure that the phone and network
load generated by registration refresh of multiple registrations do not become
excessive. This would be of particularly concern if a phone had multiple
registrations with multiple servers per registration and it is expected that some of
these servers will be unavailable.
Phone Operation for Registration

After the phone has booted up, it will register to all the servers that are
configured.
Server 1 is the primary server and supports greater SIP functionality than any
of servers. For example, SUBSCRIBE/NOTIFY services (used for features such
as shared lines, presence, and BLF) will only be established with Server 1.
Upon registration timer expiry of each server registration, the phone will
attempt to re-register. If this is unsuccessful, normal SIP re-registration
behavior (typically at intervals of 30 to 60 seconds) will proceed and continue
until the registration is successful (for example, when the Internet link is once
again operational). While the primary server registration is unavailable, the
next highest priority server in the list will serve as the working server. As soon
as the primary server registration succeeds, it will return to being the working
server.
Note If reg.x.server.y.register is set to 0, then phone will not register to that server.
However, the INVITE will fail over to that server if all higher priority servers are
down.
Recommended Practices for Fallback Deployments

In situations where server redundancy for fall-back purpose is used, the
following measures should be taken to optimize the effectiveness of the
solution:
1. Deploy an on-site DNS server to avoid long call initiation delays that can
result if the DNS server records expire.
2. Do not use OutBoundProxy configurations on the phone if the
OutBoundProxy could be unreachable when the fallback occurs.
SoundPoint IP phones can only be configured with one OutBoundProxy
per registration and all traffic for that registration will be routed through
this proxy for all servers attached to that registration. If Server 2 is not
accessible through the configured proxy, call signaling with Server 2 will
fail.
3. Avoid using too many servers as part of the redundancy configuration as
each registration will generate more traffic.
4. Educate users as to the features that will not be available when in
“fallback” operating mode.
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Presence
The Presence feature allows the phone to monitor the status of other
users/devices and allows other users to monitor it. The status of monitored
users is displayed visually and is updated in real time in the Buddies display
screen or, for speed dial entries, on the phone’s idle display. Users can block
others from monitoring their phones and are notified when a change in
monitored status occurs. Phone status changes are broadcast automatically to
monitoring phones when the user engages in calls or invokes do-not-disturb.
The user can also manually specify a state to convey, overriding, and perhaps
masking, the automatic behavior.
Note Notification when a change in monitored status occurs will be available in a

subsequent release.
The presence feature works differently when Microsoft Live Communications

Server 2005 is used as the call server. For more information, refer to the next
section, Microsoft Live Communications Server 2005 Integration.
Central XML file: <Ethernet The <bw>0</bw> (buddy watching) and <bb>0</bb> (buddy
(provisioning address>-directory. blocking) elements in the <Ethernet address>-directory.xml file
server) xml dictate the Presence aspects of directory entries.
• For more information, refer to Local Contact Directory on page
4-10.
Local Local Phone User The user can edit the directory contents. The Watch Buddy and
Interface Block Buddy fields control the buddy behavior of contacts.
Changes will be stored in the phone’s flash file system and backed up
the local copy.
Microsoft Live Communications Server 2005 Integration

SoundPoint IP, SoundStation IP, and VVX phones can used with Microsoft
Live Communications Server 2005 and Microsoft Office Communicator to
help improve business efficiencies and increase productivity and to share
ideas and information immediately with business contacts.
For instructions on changing the configuration files, refer to Configuration File
Examples on page 4-65.
Note Any contacts added through the SoundPoint IP, SoundStation IP, and VVX phone’s
buddy list will appear as a contact in Microsoft Office Communicator and Windows
Messenger.
4 - 64
Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Busy
Lamp Field on page 4-55.
Central Configuration file: Specify that support for Microsoft Live Communications Server 2005
(provisioning sip.cfg is enabled.
Specify the line/registration number used to send SUBSCRIBE for
presence.
• For more information, refer to Presence <pres/> on page A-91.
Turn the presence and messaging features on or off.
• For more information, refer to Feature <feature/> on page A-116.
Configuration file: Specify the number of line keys to assign per registration.
Specify the line/registration number which has roaming buddies
support enabled.
• For more information, refer to Roaming Buddies
<roaming_buddies/> on page A-154.
Specify the line/registration number which has roaming privacy
support enabled.
• For more information, refer to Roaming Privacy
<roaming_privacy/> on page A-154.

SoundPoint IP, SoundStation IP, and VVX phones can be deployed in two
basic methods. In the first method, Microsoft Live Communications
Server 2005 serves as the call server and the phones have a single registration.
In the second method, the phone has a primary registration to call server—that
is not Microsoft Live Communications Server (LCS)—and a secondary
registration to LCS for presence purposes.
To set up a single registration with Microsoft Live Communications Server 2005

as the call server:
b Locate the feature parameter.
c For the feature.1.name = presence attribute, set
feature.1.enabled to 1.
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d For the feature.2.name = messaging attribute, set

e Locate the voIpProt parameter.
Set the voIpProt.server.x.transport attribute to TCPpreferred or
TLS.
Your selection depends on the LCS configuration.
Note The TLS protocol is not supported on SoundPoint IP 300 and 500 phones.
f Set the voIpProt.server.x.address to the LCS address.

For example, voIpProt.server.1.address = "lcs2005.local"
g Set the voIpProt.SIP.lcs attribute to 1.
h (Optional) If SIP forking is desired, set voIpProt.SIP.ms-forking
attribute to 1.
Refer to SIP <SIP/> on page A-11.
i Save the modified sip.cfg configuration file.
2. Modify the phone1.cfg configuration file as follows:
a Open phone1.cfg in an XML editor.
b Locate the registration parameter.
c Set the reg.1.address to the LCS address.
For example, reg.1.address = "7778"
d Set the reg.1.server.y.address to the LCS server name.
e (Optional) Set the reg.1.server.y.transport attribute to
TCPpreferred or TLS.
f Set reg.1.auth.userId to the phone's LCS username.
For example, reg.1.auth.userId = "jbloggs"
g Set reg.1.auth.password to the LCS password.
For example, reg.1.auth.password = "Password2"
h Locate the roaming_buddies attribute.
i Set the roaming_buddies.reg element to 1.
Refer to Roaming Buddies <roaming_buddies/> on page A-154.
j Locate the roaming_privacy attribute.
k Set the roaming_privacy.reg element to 1.
Refer to Roaming Privacy <roaming_privacy/> on page A-154.
4 - 66
l Save the modified phone1.cfg configuration file.
To set up a dual registration with Microsoft Live Communications Server 2005 as

the presence server:
1. (Optional) Modify the sip.cfg configuration file as follows:
b Locate the feature parameter.
c For the feature.1.name = presence attribute, set
d For the feature.2.name = messaging attribute, set
e Locate the voIpProt parameter.
f If SIP forking is desired, set voIpProt.SIP.ms-forking attribute to 1.
Refer to SIP <SIP/> on page A-11.
g Save the modified sip.cfg configuration file.
2. Modify the phone1.cfg configuration file as follows:
a Open phone1.cfg in an XML editor.
b Locate the registration parameter.
c Select a registration to be used for the Microsoft Live Communications
Server 2005.
Typically, this would be 2.
d Set the reg.x.address to the LCS address.
For example, reg.2.address = "7778"
e Set the reg.x.server.y.address to the LCS server name.
f (Optional) Set the reg.2.server.y.transport attribute to
TCPpreferred or TLS.
g Set reg.x.auth.userId to the phone's LCS username.
For example, reg.2.auth.userId = "jbloggs"
h Set reg.x.auth.password to the LCS password.
For example, reg.2.auth.password = "Password2"
i Locate the roaming_buddies attribute.
4 - 67
j Set the roaming_buddies.reg element to the number corresponding

to the LCS registration.
For example, roaming_buddies.reg = 2
Refer to Roaming Buddies <roaming_buddies/> on page A-154.
k Locate the roaming_privacy attribute.
l Set the roaming_privacy.reg element to the number corresponding
to the LCS registration.
For example, roaming_privacy.reg = 2
Refer to Roaming Privacy <roaming_privacy/> on page A-154.
m Save the modified phone1.cfg configuration file.
Access URL in SIP Message

Introduced in SIP 2.2, this feature that allows information contained in
incoming SIP signaling to refer to XHTML web content that can be rendered
by the SoundPoint IP and SoundStation IP phone’s Microbrowser and the
Polycom VVX 1500 phone’s Browser.
Supporting this feature allows use of the phone’s display to provide
information before someone takes a call and while they are on a call (for
example, a SIP re-INVITE). The information accessible at the URL can be
anything that you want to have displayed.
Configuration changes can performed centrally at the boot server:

(boot server) sip.cfg • For more information, refer to Microbrowser <mb/> on page
A-119.
• Web Content Examples
• User Interface
• Signaling Changes
Web Content Examples

This feature can be used in the following circumstances:
• Call Center—Customer information

The URL provided allows the phone to access information about a
customer and display it before the agent takes the call.
4 - 68
• Call Center—Scripts for different call center groups

The phone can access a script of questions for an agent to ask a caller when
a call comes in. The script can be different for each agent group.
• Restaurant menu on a hotel phone

A guest dials a number for the restaurant and a voice indicates that the
menu is now available for viewing on the phone.
User Interface
There are three user interface aspects to this feature:
• Web content status indication
• Web content retrieval (spontaneous and on-demand)
• Settings menu item to control active versus passive behavior

Web Content Status Indication
When valid web content (validity is determined through a SIP header
parameter) is available for a SIP call, it is indicated by an icon that appears after
the call appearance status text, regardless of the call state. In the examples
shown below, a lightning bolt symbol is used to indicate that web content is
available for the displayed call appearance and the user is encouraged to press
the Select key to retrieve and display the content through the Microbrowser.
SoundPoint IP 330 Graphic Display
SoundPoint IP 550 Graphic Display
4 - 69
Web Content Retrieval

Web content is retrieved either spontaneously (active mode) or at the request
of the user (passive mode).
• Active Mode. Two methods can be used to achieve spontaneous web

content retrieval: static configuration parameters or parameters received
as part of the SIP signaling. If parameters received in the SIP signaling
conflict with the static configuration, the parameters in the SIP signaling
will take precedence.
If the phone is configured to spontaneously retrieve web content, the
phone will launch the interactive Microbrowser and have it fetch the
appropriate URL upon arrival of the appropriate SIP signaling, subject to
some conditions described below.
Since new web content URLs can be received at any time—as the first URL
for a call or a replacement URL—rules are needed to match displayed web
content with automatic phone behavior, which are valid actions from
within the Microbrowser context.
Spontaneous web content will only be retrieved and displayed for a call if
that call occupies, or will occupy, the UI focus at the time of the event.
• Passive Mode. Web content can also be retrieved when the user chooses
to do so. The fact that web content is available for viewing is shown
through the call appearance-based web content icon described in Web
Content Status Indication on page 4-69. The Select key can be used to fetch
the associated web content for the call that is in focus. If the web content
has expired, the icon will be removed and the Select key will perform no
function.
Passive mode is recommended for applications where the Microbrowser
is used for other applications. In the SIP 2.2 feature, interactive
microbrowser sessions will be interrupted by the arrival of active-mode
web content URLs, which may cause annoyance, although the Back
navigation function will work in this context.
Settings Menu
If enabled, a new SIP web content entry is added to the Setting > Basic >
Preferences menu to allow the user to change the current content retrieval
mode. Two options are provided: passive mode and active mode.
Signaling Changes
A new SIP header must be used to report web content associated with SIP
phone calls (the SSAWC header follow the BNF for the standard SIP header
Alert-Info):
Alert-Info = "Alert-Info" HCOLON alert-param *(COMMA alert-param)

alert-param = LAQUOT absoluteURI RAQUOT *( SEMI generic-param )
4 - 70
The web content must be located with an absolute URI, which begins with the
scheme identifier. Currently only the HTTP scheme is supported.
So an example header might look like:
Access-URL: <http://server.polycom.com/content23456.xhtml>
This header may be placed in SIP requests and responses, as appropriate so

long as the messages are part of an INVITE-initiated dialog and the phone can
associate them with an existing phone call.
This feature also requires the definition of two optional parameters:
• An expires parameter is defined to indicate the lifespan of the URL itself,

or, assuming that the URL is permanent, the time span for which the
content is expected to have relevance to the call with which it is associated.
If the parameter is absent or invalid, this will be interpreted to mean that
the content or the URL itself will be persistent in nature. A value, if it is
present, will indicate the lifespan of the content in seconds (zero has
special significance—see example below). When the lifespan expires, the
phone will remove both the indication of the URL and the ability of the
user to retrieve it.
For example:
Access-URL:
<http://server.polycom.com/content23456.xhtml>;expires=60
If the server wishes to invalidate a previous URL, it can send a new header
(through UPDATE) with expires=0. The expires parameter is ignored when
determining whether to spontaneously retrieve the web content unless
expires=0.
• A mode parameter is defined to indicate whether the web content should

be displayed spontaneously or retrieved on-demand. Two values are
allowed: active and passive. If the parameter is absent or invalid, this will
be interpreted the same as passive, meaning that the web content will be
retrievable on-demand but will not be spontaneously displayed. If the
value is set to active, the web content will be spontaneously displayed,
subject to the rules discussed under Active Mode in Web Content
Retrieval on page 4-70.
For example:
Access-URL:
<http://server.polycom.com/content23456.xhtml>;expires=60;mode
=passive
In this case, the phone will indicate in the call appearance user interface
that web content is available for a period of 60 seconds and will retrieve
the web content at the request of the user for a period of up to 60 seconds
but the phone will not spontaneously switch to the microbrowser
application and download the content.
4 - 71
Static DNS Cache

Starting with SIP 2.1.0, failover redundancy can only be utilized when the
configured IP server hostname resolves (through SRV or A record) to multiple
IP addresses. Unfortunately, some customer’s are unable to configure the DNS
to take advantage of failover redundancy.
The solution in SIP 3.1 is to provide the ability to statically configure a set of
DNS NAPTR SRV and/or A records into the phone.
When a phone is configured with a DNS server, it will behave as follows by
default:
• An initial attempt to resolve a hostname that is within the static DNS

cache, for example to register with its SIP registrar, results in a query to the
DNS.
• If the initial DNS query returns no results for the hostname or cannot be
contacted, then the values in the static cache are used for their configured
time interval.
• After the configured time interval has elapsed, a resolution attempt of the
hostname will again result in a query to the DNS.
• If a DNS query for a hostname that is in the static cache returns a result,
the values from the DNS are used and the statically cached values are
ignored.
When a phone is not configured with a DNS server, it will behave as follows
• An attempt to resolve a hostname that is within the static DNS cache will
always return the results from the static cache.
Support for negative DNS caching as described in RFC 2308 is also provided
to allow faster failover when prior DNS queries have returned no results from
the DNS server. For more information, go to
http://tools.ietf.org/html/rfc2308 .
Central Configuration file: Specify DNS NAPTR, SRV, and A records for use when the phone is
(boot server) sip.cfg not configured to use a DNS server.
• For more information, refer to DNS Cache <dns/> on page A-126.
Polycom recommends that you create another file with your organization’s
modifications. If you must change any Polycom templates, back them up first.
4 - 72
Example 1
This example shows how to configure static DNS cache using A records IP
addresses in SIP server address fields.
When the static DNS cache is not used, the sip.cfg configuration would look
as follows:
reg.1.address="1001"
reg.1.server.1.address="172.23.0.140"
reg.1.server.1.port="5075"
reg.1.server.1.transport="UDPOnly"
When the static DNS cache is used, the sip.cfg configuration would look as
follows:
reg.1.server.1.address="sipserver.example.com"
reg.1.server.2.address=""
reg.1.server.2.port=""
reg.1.server.2.transport=""
dns.cache.A.1.name="sipserver.example.com"
dns.cache.A.1.ttl="3600"
dns.cache.A.1.address="172.23.0.140"
dns.cache.A.2.name="sipserver.example.com"
Note Above addresses are presented to SIP application in order, for example,
dns.cache.A.1, dns.cache.A.2, and so on.
Example 2
This example shows how to configure static DNS cache where your DNS
provides A records for server.X.address but not SRV. In this case, the static
DNS cache on the phone provides SRV records. For more information, go to
as follows:
reg.1.address="1002@sipserver.example.com"
reg.1.server.1.address="primary.sipserver.example.com"
reg.1.server.2.address="secondary.sipserver.example.com"
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follows:
dns.cache.SRV.1.name="_sip._udp.sipserver.example.com "
dns.cache.SRV.1.ttl= "3600"
dns.cache.SRV.1.priority="1"
dns.cache.SRV.1.weight="1"
dns.cache.SRV.1.port="5075"
dns.cache.SRV.1.target="primary.sipserver.example.com"
dns.cache.SRV.2.target="secondary.sipserver.example.com
Note The reg.1.server.1.port and reg.1.server.2.port values in this example are

set to null to force SRV lookups.
Example 3
This example shows how to configure static DNS cache where your DNS
provides NAPTR and SRV records for server.X.address .
as follows:
reg.1.address="1002@sipserver.example.com
follows:
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dns.cache.NAPTR.1.name="sipserver.example.com"
dns.cache.NAPTR.1.ttl= "3600"
dns.cache.NAPTR.1.order="1"
dns.cache.NAPTR.1.preference="1"
dns.cache.NAPTR.1.flag="s"
dns.cache.NAPTR.1.service=" SIP+D2U"
dns.cache.NAPTR.1.regexp=""
dns.cache.NAPTR.1.replacement="_sip._udp.sipserver.example.com"
dns.cache.SRV.1.target="primary.sipserver.example.com"
dns.cache.SRV.2.target="secondary.sipserver.example.com
dns.cache.A.1.name="primary.sipserver.example.com"
dns.cache.A.2.name="secondary.sipserver.example.com"
Note The reg.1.server.1.port, reg.1.server.2.port,

reg.1.server.1.transport, and reg.1.server.2.transport values in this
example are set to null to force NAPTR lookups.
Display of Warnings from SIP Headers

The Warning Field from a SIP header may be used to cause the phone to
display a three second “pop-up” to the user. For example, this feature can be
used to inform the user of information such as the reason that a call transfer
action failed (bad extension entered, for example). (For more information,
refer to Header Support on page B-4.)
4 - 75
These messages are displayed in any language supported by the phone for
three seconds unless overridden by another message or action.
For example, if a user parks a call, the following message appears on their
phone:
Central Configuration file: Turn this feature on or off and specify which warnings are
(boot server) sip.cfg displayable.
Quick Setup of SoundPoint IP / SoundStation IP / VVX Phones

In the SIP 3.1.2 release, a Quick Setup feature was added to simplify the
process of entering the provisioning (boot) server parameters from the phone’s
user interface. This feature is designed to make it easier for on-site, “out of the
box” provisioning of SoundPoint IP, SoundStation IP, and VVX phones.
When enabled, this feature will present a QSetup soft key to the user. When
the user presses the QSetup soft key, a new menu will immediately appear
that allows them to configure the necessary parameters for the phone to access
the provisioning server for configuration. The QSetup soft key may be
disabled using a configuration file setting such that it does not appear after it
has been successfully configured.
The Quick Setup feature is supported on all SoundPoint IP 32x/33x, 430, 450,
550, 560, 650, and 670 desktop phones, SoundStation IP 6000 and 7000
conference phones, and Polycom VVX 1500 phones.
System administrators can enable the Quick Setup feature through the use of
a new parameter in sip.cfg configuration file (or through the phone’s menu).
For details on how to configure SoundPoint IP, SoundStation IP, and VVX
phones for quick setup, refer to “Technical Bulletin 45460: Using Quick Setup
with SoundPoint IP, SoundStation IP, and Polycom VVX 1500 Phones” at
4 - 76
Central Configuration file: Turn this feature on or off .

(boot server) sip.cfg • For more information, refer to Provisioning <prov/> on page
A-114.
Setting Up Audio Features

Proprietary state-of-the-art digital signal processing (DSP) technology is used
to provide an excellent audio experience.
following audio-related features:
• Low-Delay Audio Packet Transmission
• Jitter Buffer and Packet Error Concealment
• Voice Activity Detection
• DTMF Tone Generation
• DTMF Event RTP Payload
• Acoustic Echo Cancellation
• Audio Codecs
• Background Noise Suppression
• Comfort Noise Fill
• Automatic Gain Control
• IP Type-of-Service
• IEEE 802.1p/Q
• Voice Quality Monitoring
• Dynamic Noise Reduction
• Treble/Bass Controls
Low-Delay Audio Packet Transmission

The phone is designed to minimize latency for audio packet transmission.
4 - 77
Jitter Buffer and Packet Error Concealment

The phone employs a high-performance jitter buffer and packet error
concealment system designed to mitigate packet inter-arrival jitter and
out-of-order or lost (lost or excessively delayed by the network) packets. The
jitter buffer is adaptive and configurable for different network environments.
When packets are lost, a concealment algorithm minimizes the resulting
negative audio consequences.
or locally:
Central Configuration file: Set the jitter buffer tuning parameters including minimum and
(provisioning sip.cfg maximum size and shrink aggression.
server) • For more information, refer to Codec Profiles <audioProfile/> on
page A-49.
Local Web Server Set the jitter buffer tuning parameters including minimum and
(if enabled) maximum size and shrink aggression.
Navigate to http://<phoneIPAddress>/coreConf.htm#au
Voice Activity Detection

The purpose of voice activity detection (VAD) is to conserve network
bandwidth by detecting periods of relative “silence” in the transmit data path
and replacing that silence efficiently with special packets that indicate silence
is occurring. For those compression algorithms without an inherent VAD
function, such as G.711, the phone is compatible with the comprehensive
codec-independent comfort noise transmission algorithm specified in RFC
3389. This algorithm is derived from G.711 Appendix II, which defines a
comfort noise (CN) payload format (or bit-stream) for G.711 use in
packet-based, multimedia communication systems. The phone generates CN
packets (also known as Silence Insertion Descriptor (SID) frames) and also
decodes CN packets, efficiently regenerating a facsimile of the background
noise at the remote end.
Central Configuration file: Enable or disable VAD and set the detection threshold.
(provisioning sip.cfg • For more information, refer to Voice Activity Detection <vad/> on
server) page A-60.
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DTMF Tone Generation

The phone generates dual tone multi-frequency (DTMF) tones in response to
user dialing on the dial pad. These tones are transmitted in the real-time
transport protocol (RTP) streams of connected calls. The phone can encode the
DTMF tones using the active voice codec or using RFC 2833 compatible
encoding. The coding format decision is based on the capabilities of the remote
end point.
Central Configuration file: Set the DTMF tone levels, autodialing on and off times, and other
(provisioning sip.cfg parameters.
server) • For more information, refer to Dual Tone Multi-Frequency
<DTMF/> on page A-35.
DTMF Event RTP Payload

The phone is compatible with RFC 2833 - RTP Payload for DTMF Digits,
Telephony Tones, and Telephony Signals. RFC 2833 describes a standard
RTP-compatible technique for conveying DTMF dialing and other telephony
events over an RTP media stream. The phone generates RFC 2833 (DTMF
only) events but does not regenerate, nor otherwise use, DTMF events
received from the remote end of the call.
Central Configuration file: Enable or disable RFC 2833 support in SDP offers and specify the
(provisioning sip.cfg payload value to use in SDP offers.
server) • For more information, refer to Dual Tone Multi-Frequency
<DTMF/> on page A-35.
Acoustic Echo Cancellation

The phone employs advanced acoustic echo cancellation (AEC) for hands-free
operation. Both linear and non-linear techniques are employed to aggressively
reduce echo yet provide for natural full-duplex communication patterns.
When using the handset on any SoundPoint IP phones, AEC is not normally
required. In certain situations, where echo is experienced by the far-end party,
when the user is on the handset, AEC may be enabled to reduce/avoid this
echo. To achieve this, make the following changes in the sip.cfg configuration
file (default settings for these parameters are disabled):
voice.aec.hs.enable = 1
voice.aes.hs.enable = 1
voice.ns.hs.enable = 1
voice.ns.hs.signalAttn = -6
voice.ns.hs.silenceAttn = -9
4 - 79
For more information, refer to Acoustic Echo Cancellation <aec/> on page

A-44, Acoustic Echo Suppression <aes/> on page A-55, and Background
Noise Suppression <ns/> on page A-56.
Audio Codecs
The following table shows which audio codecs are support by each of the
SoundPoint IP, SoundStation IP, and Polycom VVX phones:
Phone Supported Audio Codecs
SoundPoint IP 430 G.711μ-law, G.711a-law, G.729AB
SoundPoint IP 320/321/330/331 G.711μ-law, G.711a-law, G.729AB, iLBC
SoundPoint IP 335, 450, 550, 560, 650, and 670 G.711μ-law, G.711a-law, G.722, G.729AB, iLBC
SoundStation IP 6000 G.711μ-law, G.711a-law, G.722, G.722.1, G.722.1C,

G.729AB, Siren14, iLBC
SoundStation IP 7000 G.711μ-law, G.711a-law, G.722, G.722.1, G.722.1C,

G.729AB, Lin16, Siren14, Siren22, iLBC
Polycom VVX 1500 G.711μ-law, G.711a-law, G.719, G.722, G.722.1,

G.722.1C, G.729AB, Lin16, Siren14, iLBC
The following table summarizes the supported audio codecs:
Effective
Sample audio
Algorithm MIME Type Ref. Bit Rate Rate Frame Size bandwidth
G.711μ-law PMCU RFC 1890 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz
G.711a-law PCMA RFC 1890 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz
G.719 G719/48000 RFC 5404 32 Kbps, 48 kHz 20ms 20kHz

48 Kbps,
64 Kbps
G.722 G722/8000 RFC 1890 64 Kbps 16 Ksps 10ms - 80ms 7 KHz
G.722.1 G7221/16000 RFC 3047 16 Kbps, 16 Ksps 20ms - 80ms 7 KHz

24 Kbps,
32 Kbps
G.722.1C G7221/ G7221C 24 Kbps 32 Ksps 20ms - 80ms 14 KHz

32000 32 Kbps
48 Kbps
G.729AB G729 RFC 1890 8 Kbps 8 Ksps 10ms - 80ms 3.5KHz
SID CN RFC 3389 N/A N/A N/A N/A
4 - 80
Effective
Sample audio
Algorithm MIME Type Ref. Bit Rate Rate Frame Size bandwidth
Lin16 L16/8000 RFC 1890 128 Kbps 8 Ksps 10ms 3.5 KHz
L16/16000 256 Kbps 16 Ksps 7 KHz
L16/32000 512 Kbps 32 Ksps 14 KHz
L16/44100 705.6 Kbps 44.1 Ksps 20 KHz
L16/48000 768 Kbps 48 Ksps 22 KHz
Siren14 SIREN14/ SIREN14 24 Kbps 32 Ksps 20ms - 80ms 14 KHz

16000 32 Kbps
48 Kbps
Siren22 SIREN22/ SIREN22 32 Kbps 48 Ksps 20ms - 80ms 22 KHz

48000 48 Kbps
64 Kbps
RFC 2833 phone-event RFC 2833 N/A N/A N/A N/A
iLBC iLBC RFC 3951 13.33Kbps 8 Ksps 30ms - 60ms 3.5KHz

15.2Kbps 20ms - 80ms
Note The network bandwidth necessary to send the encoded voice is typically 5-10%
higher than the encoded bit rate due to packetization overhead. For example, a
G.722.1C call at 48kbps consumes about 100kbps of network bandwidth (two-way
audio).

or locally:
Central Configuration file: Specify codec priority, preferred payload sizes, and jitter buffer tuning
server) • For more information, refer to Codec Preferences <codecPref/>
on page A-45 and Codec Profiles <audioProfile/> on page A-49.
Local Web Server Specify codec priority, preferred payload sizes, and jitter buffer tuning
(if enabled) parameters.
Navigate to http://<phoneIPAddress>/coreConf.htm#au
Background Noise Suppression

Background noise suppression (BNS) is designed primarily for hands-free
operation and reduces background noise to enhance communication in noisy
environments.
4 - 81
Comfort Noise Fill

Comfort noise fill is designed to help provide a consistent noise level to the
remote user of a hands-free call. Fluctuations in perceived background noise
levels are an undesirable side effect of the non-linear component of most AEC
systems. This feature uses noise synthesis techniques to smooth out the noise
level in the direction toward the remote user, providing a more natural call
experience.
Automatic Gain Control

Automatic Gain Control (AGC) is applicable to hands-free operation and is
used to boost the transmit gain of the local talker in certain circumstances. This
increases the effective user-phone radius and helps with the intelligibility of
soft-talkers.
IP Type-of-Service
The “type of service” field in an IP packet header consists of four
type-of-service (TOS) bits and a 3-bit precedence field. Each TOS bit can be set
to either 0 or 1. The precedence field can be set to a value from 0 through 7. The
type of service can be configured specifically for RTP packets and call control
packets, such as SIP signaling packets.
or locally:
Central Configuration file: Specify protocol-specific IP TOS settings.

(provisioning sip.cfg • For more information, refer to IP TOS <IP/> on page A-72.
server)
Local Web Server Specify IP TOS settings.

(if enabled) Navigate to: http://<phoneIPAddress>/netConf.htm#qo
IEEE 802.1p/Q
The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN
header for one of the following reasons:
• When it has a valid VLAN ID set in its network configuration
4 - 82
• When it is instructed to tag packets through Cisco Discovery Protocol

(CDP) running on a connected Ethernet switch
• When a VLAN ID is obtained from DHCP (refer to DHCP Menu on page

3-8)
The 802.1p/Q user_priority field can be set to a value from 0 to 7. The
user_priority can be configured specifically for RTP packets and call control
packets, such as SIP signaling packets, with default settings configurable for
all other packets.
or locally:
Central Configuration file: Specify default and protocol-specific 802.1p/Q settings.

(provisioning sip.cfg • For more information, refer to Ethernet IEEE 802.1p/Q
server) <ethernet/> on page A-71.
Local Web Server Specify 802.1p/Q settings.

(if enabled) Navigate to http://<phoneIPAddress>/netConf.htm#qo
Local Phone User Specify whether CDP is to be used or manually set the VLAN ID or
Interface configure DHCP VLAN Discovery.
Phase 1: BootRom - Navigate to: SETUP menu during auto-boot
countdown.
Phase 2: SIP Application - Navigate to:
Menu>Settings>Advanced>Admin Settings>Network
Configuration
• For more information, refer to Setting Up the Network on page
3-2.
Voice Quality Monitoring
Note This feature requires a license key for activation except for the Polycom VVX 1500.
Using this feature may require purchase of a license key or activation by Polycom
The SoundPoint IP phones can be configured to generate various quality

metrics for listening and conversational quality. These metrics can be sent
between the phones in RTCP XR packets. The metrics can also be sent as SIP
PUBLISH messages to a central voice quality report collector. The collection of
these metrics is supported on the SoundPoint IP 32x/33x, 430, 450, 550, 560,
650, and 670 phones and the Polycom VVX 1500 phone.
Note Voice Quality Monitoring is not supported on the SoundStation IP 6000 and 7000
conference phones at this time. Only Voice Quality Monitoring of the audio portion
is supported on the Polycom VVX 1500 at this time.
4 - 83
The RTCP XR packets are compliant with RFC 3611 - RTP Control Extended
Reports (RTCP XR). The packets are sent to a report collector as specified in
draft RFC draft-ietf_sipping_rtcp-summary-02.
Three types of quality reports can be enabled:
• Alert—Generated when the call quality degrades below a configurable

threshold.
• Periodic—Generated during a call at a configurable period.
• Session—Generated at the end of a call.

A wide range of performance metrics are generated. Some are based on
current values, such as jitter buffer nominal delay and round trip delay, while
others cover the time period from the beginning of the call until the report is
sent, such as network packet loss. Some metrics are computed using other
metrics as input, such as listening Mean Opinion Score (MOS), conversational
MOS, listening R-factor, and conversational R-factor.
Central Configuration file: Specify the location of the central report collector, how often the
(provisioning sip.cfg reports are generated, and the warning and critical threshold values
server) that will cause generation of alert reports.
• For more information, refer to Quality Monitoring <quality
monitoring/> on page A-61.
Dynamic Noise Reduction

Dynamic noise reduction (DNR) provides maximum microphone sensitivity,
while automatically reducing background noise— from fans, projectors,
heating and air conditioning—for clearer sound and more efficient
conferencing.
Treble/Bass Controls
The treble and bass controls equalize the tone of the high and low frequency
sound from the speakers.
The SoundStation IP 7000 phone’s treble and bass controls can be modified by
the user (through Menu > Settings > Basic > Audio > Treble EQ or Bass EQ).
Central Configuration file: Specify the user’s preferences for treble and bass.
server) A-31.
4 - 84
Setting Up Video Features

The Polycom VVX 1500 phone supports transmission and reception of high
quality video images. The video is compatible with RFC 3984 - RTP Payload
Format for H.264 Video, RFC 4629 - RTP Payload Format for ITU-T Rec. H.263
Video, and RFC 5168 - XML Schema for Media Control.
following video-related features:
• Video Transmission
• Video Codecs
• H.323 Protocol
Video Transmission
By default, at the start of a video call, the Polycom VVX 1500 phone transmits
an RTP encapsulated video stream with images captured from the local
camera. Users can stop and start video transmission by pressing the Video
key, and then selecting the appropriate soft key.
You can control of the following features of the Polycom VVX 1500 phone’s
camera:
• Flicker avoidance
• Frame rate
• Brightness level
• Saturation level
• Contrast level
• Sharpness level
4 - 85

or locally:
Central Configuration file: Turn video transmission off at the near end when calls start and
(provisioning sip.cfg transmit still image if video not available.
server) • For more information, refer to Video Settings <video/> on page
A-64.
Specify camera parameters.

• For more information, refer to Camera Controls <camera/> on
page A-70.
Determine how the local camera is displayed.
• For more information, refer to Local Camera View
<localCameraView/> on page A-71.
Local Local Phone User The user can set the individual video settings from the menu through
Interface Settings > Basic > Video > Video Call Settings, Video Screen
Mode, and Local Camera View.
The user can set the individual camera settings from the menu
through Settings > Basic > Video > Camera Settings.
Video Codecs
The following table summarizes the Polycom VVX 1500 phone’s video codec
support:
Effective
Frame video
Algorithm MIME Type Bit Rate Rate Frame Size bandwidth
H.261 H261/90000 64 kbps to 5 fps to Tx Frame size: CIF, Refer to Bit

768 kbps 30 fps QCIF Rate column.
RX Frame size: CIF,
QCIF
H.263 H263/90000, Tx Frame size:CIF,

H263-1998/90000, QCIF
H263-2000/90000 Rx Frame size:CIF,
QCIF, SQCIF, QVGA,
H.264 H264/90000 SVGA, SIF
Central Configuration file: Specify codec priority, payload type and jitter buffer tuning
server) • For more information, refer to Codec Preferences <codecPref/>
on page A-66 and Codec Profiles <profile/> on page A-66.
4 - 86
H.323 Protocol
Note This feature requires a license key for activation on the Polycom VVX 1500. Using
this feature may require purchase of a license key or activation by Polycom
As of SIP 3.2.2, telephony signaling support via the H.323 family of protocols
enabling direct communication with H.323 endpoints, gatekeepers, call and
media servers, and signaling gateways is supported on the Polycom VVX 1500
phone.
SIP and H.323 signaling can be supported at the same time, including bridging
both types of calls during multi-party conference calls. Automatic detection of
the correct or optimal signaling protocol is available when dialing from the
contact or corporate directories. While SIP supports server redundancy and
several transport options, only a single configured H.323 gatekeeper address
per phone is supported. H.323 gatekeepers are optional, but if available, they
must be used. If a gatekeeper is not configured or unavailable, calls can still be
made if so configured.
If the H.323 protocol is disabled, there will be no visible evidence in the user
interface of the Polycom VVX 1500 phone.
Support of the SIP protocol for telephony signaling can be disabled on the
Polycom VVX 1500 such that all calls would be routed via the H.323 protocol.
• Supported Standards
• Supported Polycom Interoperability
• Configuration File Changes
• Useful Tips
• Examples
Supported Standards
The following standards are supported by the implementation of this feature:
Standard Description
ITU-T Recommendation H.323 (2003) Packet-based multimedia

communications systems
ITU-T Recommendation Q.931 (1998) ISDN user-network interface layer 3

specification for basic call control
ITU-T Recommendation H.225.0 (2003) Call signaling protocols and media

stream packetization for packet based
multimedia communications systems
4 - 87
Standard Description
ITU-T Recommendation H.245 (5/2003) Control protocol for multimedia

communication
ITU-T Recommendation H.235.0 - Security and encryption for H Series

H.235.9 (2005) (H.323 and other H.245 based)
multimedia terminals
ITU-T Recommendation H.350.1 Directory services architecture for

(8/2003) H.323
Supported Polycom Interoperability

Video calls are supported to the following Polycom endpoints/bridges/call
servers (or gatekeepers)/media servers:
Make/Model Protocol Software Version
Polycom RMX 1000™ H.323 SW 1.1.1.8787
Polycom RMX 2000™ H.323 SW 4.0.2.7
Polycom HDX 9000™ series SIP/ISDN/H.323 SW 2.5.0.6
Polycom HDX 8000™ SIP/ISDN/H.323 SW 2.5.0.6
Polycom VSX™ 8000 SIP/ISDN/H.323 SW 9.0.5.1
Polycom VSX™ 7000s and VSX™ SIP/ISDN/H.323 SW 9.0.5.1

7000e
Polycom VSX™ 6000 and 6000a SIP/ISDN/H.323 SW 9.0.5.1
Polycom V700™ SIP/ISDN/H.323 SW 9.0.5.1
Polycom V500™ SIP/ISDN/H.323 SW 9.0.5.1
Note Refer to the Release Notes for the latest list of supported Polycom
endpoints/bridges/call servers (or gatekeepers)/media servers and any supported
third party products. Any issues (and possible workarounds) with any of the
above-mentioned products are also documented in the Release Notes.
4 - 88

Central Configuration file: Specify H.323 protocol, server, gatekeeper, and DTMF signaling
server) • For more information, refer to Protocol <voIpProt/> on page A-7.
Specify call routing parameters.
A-31 and Call Handling Configuration <call/> on page A-80.
Specify the video call rate.
• For more information, refer to Video Settings <video/> on page
A-64.
Specify auto-answer parameters.
on page A-80.
Specify H.323 media encryption parameters.
• For more information, refer to H.235 <H235/> on page A-112.
Configuration file: Specify the registration protocol.

Specify the auto off-hook protocol for dual-protocol lines.
• For more information, refer to Calls <call/> on page A-139.
Useful Tips
The following information should be noted:
• If the phone has only the H.323 protocol enabled, it cannot be used to
answer SIP calls.
• If the phone has only the SIP protocol enabled, it cannot be used to answer
H.323 calls.
• If both SIP and H.323 protocols are disabled by mistake, the phone will
continue to work as a SIP-only phone; however, the phone is not
registered (you are able to send and receive SIP URL calls).
• The protocol that is used to place a call is stored in the placed call list of the
user’s phone.
• The protocol to be used when placing a call from the user’s local contact
directory is unspecified by default. The user can select SIP or H.323.
• The protocol that is used when placing a call from the user’s corporate
directory depends on the order of the attributes in the corporate directory.
If only SIP_address is defined, then the SIP protocol is used. If only
H323_address is defined, then the H.323 protocol is used. If both are
defined, then the one that is defined first is used. For example, if
dir.corp.attribute.4.type is SIP_address and
4 - 89
dir.corp.attribute.5.type is H323_address, then the SIP protocol is

used.
• By default, the user is presented with protocol routing choices when a call
could be placed with more than one protocol from its current context. The
user must choose between SIP and H.323 to place a call by pressing the
appropriate soft key.
• Calls made using H.323 cannot be forwarded or transferred.

— The Transfer and Forward soft keys are not displayed during an
H.323 call on a Polycom VVX 1500 phone. The Forward soft key is not
displayed on the idle display of a Polycom VVX 1500 phone if the
primary line is an H.323 line.
— If a Polycom VVX 1500 user presses the Transfer soft key during an
H.323 call on a Polycom VVX 1500 phone, no action is taken.
— The auto-divert field in the local contact directory entry is ignored
when a call is placed to that contact using H.323.
— If a conference host ends a three-way conference the call and one of the
party is connected by H.323, that party is not transferred to the other
party.
Examples
The following example sip.cfg configuration file shows the relevant
parameters:
• To configure both SIP and H.323 protocols
• To set up a SIP and H.323 dial plan—Numbers with the format “0xxx” are
placed on a SIP line and numbers with the format “33xx” are placed on an
H.323 line
• To set up manual protocol routing using soft keys—If the protocol to use
to place a call cannot be determined, the Use SIP and Use H.323 soft keys
appear, and the user must select one for the call to be placed.
• To configure auto-answering on H.323 calls only
• To set the preferred protocol to SIP
4 - 90
The following example phone1.cfg configuration file shows the relevant

parameters:
• To set to configure one SIP line, one H.323 line, and a dual protocol
line—both SIP and H.323 can be used.
• To set the preferred protocol for off-hook calls on the third (dual protocol)
line to SIP
4 - 91
Setting Up Security Features

following security-related features:
• Local User and Administrator Privilege Levels
• Custom Certificates
• Incoming Signaling Validation
• Secure Real-Time Transport Protocol
• Configuration File Encryption
• Digital Certificates
• Mutual TLS Authentication
Local User and Administrator Privilege Levels

Several local settings menus are protected with two privilege levels, user and
administrator, each with its own password. The phone will prompt for either
the user or administrator password before granting access to the various menu
options. When the user password is requested, the administrator password
will also work. The web server is protected by the administrator password
(refer to Configuring SoundPoint IP / SoundStation IP / VVX Phones Locally
on page 4-98).
or locally:
Central Configuration file: Specify the minimum lengths for the user and administrator
(provisioning sip.cfg passwords.
server) • For more information, refer to Password Lengths
<pwd/><length/> on page A-109.
Local Web Server None.

(if enabled)
Local Phone User The user and administrator passwords can be changed under the
Interface Settings menu or through configuration parameters (refer to Flash
Parameter Configuration on page A-157). Passwords can consist of
ASCII characters 32-127 (0x20-0x7F) only.
Changes are saved to local flash but are not backed up to <Ethernet
address>-phone.cfg on the provisioning server for security reasons.
4 - 92
Custom Certificates
The phone trusts certificates issued by widely recognized certificate
authorities when trying to establish a connection to a provisioning server for
application provisioning. Refer to Trusted Certificate Authority List on page
C-1.
In addition, custom certificates can be added to the phone. This is done by
using the SSL Security menu on the phone to provide the URL of the custom
certificate then select an option to use this custom certificate.
Note For more information on using custom certificates, refer to “Technical Bulletin
17877: Using Custom Certificates With SoundPoint IP Phones” at
http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pu
b.html .
Configuration changes can be performed locally:
Local Local Phone User The custom certificate can be specified and the type of certificate to
Interface trust can be set under the Settings menu.
Incoming Signaling Validation

The three optional levels of security for validating incoming network signaling
are:
• Source IP address validation
• Digest authentication
• Source IP address validation and digest authentication

Central Configuration File: Specify the type of validation to perform on a request-by-request

(provisioning sip.cfg basis, appropriate to specific event types in some cases.
server) • For more information, refer to Request Validation
<requestValidation/> on page A-19.
Secure Real-Time Transport Protocol

Secure Real-Time Transport Protocol (SRTP) provides means of encrypting the
audio stream(s) of VoIP phone calls to avoid interception and eavesdropping
on phone calls. Both RTP and RTCP signaling may be encrypted using an AES
algorithm as described in RFC3711. When this feature is enabled, phones will
negotiate with the other end-point whether and what type of encryption or
4 - 93
authentication to utilize for the session. This negotiation process is compliant

with RFC4568 (Session Description Protocol (SDP) Security Descriptions for
Media Streams).
Authentication proves to the phone receiving the RTP/RTCP stream that the
packets are from the expected source and have not been tampered with.
Encryption modifies the data in the RTP/RTCP streams so that, if the data is
captured or intercepted, it cannot be understood—it sounds like noise. Only
the receiver knows the key to restore the data.
A number of configuration parameters have been added to allow you to turn
off authentication and encryption for RTP and RTCP streams. This is done
mainly to allow the system administrator to reduce the CPU usage on “legacy”
Polycom phones in certain deployment scenarios (for example, if three-way
local conferencing is required).
Note When using SRTP with Polycom VVX 1500 phone, limit the video bandwidth on the
Polycom VVX 1500 to 384kbps; otherwise you will experience performance issues.
If the call is completely secure (RTP authentication and encryption and RTCP
authentication and RTCP encryption are enabled), then the user sees a padlock
symbol appearing in the last frame of the connected context animation
(two arrow moving towards each other).
In SIP 2.2, the SRTP feature has been implemented in a very configurable
manner. The reason for this is to allow deployment in a mixed environment
where some elements of the solution are SRTP capable and some are not.
In SIP 3.2, sec.srtp.requireMatchingTag was added to sip.cfg as a flag to
force a check of the tag value in the crypto attribute in an SDP answer.
For detailed configuration instructions, refer to “Technical Bulletin 25751:
Secure Real-Time Transport Protocol on SoundPoint IP Phones” at
Central Configuration File: Specify the parameters to enable and disable SRTP.
(boot server) sip.cfg • For more information, refer to SRTP <srtp/>on page A-110.
4 - 94
Configuration File Encryption

Configuration files (excluding the master configuration file), contact
directories, and configuration override files can all be encrypted.
Note Encrypted configuration files can be decrypted on the SoundPoint IP 32x, 33x, 430,
450, 550, 560, 650, and 670, the SoundStation IP 6000, and 7000, and the
Polycom VVX 1500 phones.
The master configuration file cannot be encrypted on the provisioning server. This
file is downloaded by the BootROM that does not recognize encrypted files. For
more information, refer to Master Configuration Files on page A-2.
For more information on encrypting configuration files including determining

whether an encrypted file is the same as an unencrypted file and using the
SDK to facilitate key generation, refer to Encrypting Configuration Files on
page C-4.
Central Configuration File: Specify the phone-specific contact directory and the
(provisioning sip.cfg phone-specific configuration override file.
server) • For more information, refer to Encryption <encryption/>
on page A-109.
Configuration file: Change the encryption key.

<device>.cfg • For more information, refer to refer to Flash Parameter
Digital Certificates
Starting in May 2009, Polycom is installing a digital certificate on certain
SoundPoint IP phone models at the manufacturing facility. Over time, other
SoundPoint IP phone models as well as all SoundStation IP and Polycom VVX
phone models will have a digital certificate. Refer to “Technical Bulletin 37148:
Device Certificates on SoundPoint IP, SoundStation IP, and Polycom VVX
1500 Phones“at
This X.509 digital certificate is ‘signed’ by the Polycom Root CA and may be
used for a server to authenticate the phone when initiating Transport Layer
Security (TLS) based communications such as those used for HTTPS
provisioning and TLS SIP signaling encryption. The Polycom Root CA can be
downloaded from http://pki.polycom.com/pki . The X.509 digital certificates
are set to expire on March 9, 2044.
4 - 95
An X.509 digital certificate is a digitally signed statement. The X.509 standard

defines what information can go into a certificate. All X.509 certificates have
the following fields, in addition to the signature:
• Version. This identifies which version of the X.509 standard applies to this
certificate, which affects what information can be specified in it.
• Serial Number. The entity that created the certificate is responsible for
assigning it a serial number to distinguish it from other certificates it
issues.
• Signature Algorithm Identifier. This identifies the algorithm used by the

Certificate Authority (CA) to sign the certificate.
• Issuer Name. The X.500 name of the entity that signed the certificate. This
is normally a CA. Using this certificate implies trusting the entity that
signed this certificate.
• Validity Period. Each certificate is valid only for a limited amount of time.
This period is described by a start date and time and an end date and time,
and can be as short as a few seconds or almost as long as a century.
• Subject Name. The name of the entity whose public key the certificate
identifies. This name uses the X.500 standard, so it is intended to be unique
across the Internet.
• Subject Public Key Information. This is the public key of the entity being
named, together with an algorithm identifier which specifies which public
key cryptographic system this key belongs to and any associated key
parameters.
The following is an example of a Polycom device certificate (if opened with the
Microsoft Internet Explorer 7 or Firefox 3.5 browser on a computer running
Microsoft XP Service Pack 3):
The device certificate and associated private key are stored on the phone in its
non-volatile memory as part of the manufacturing process.
4 - 96
For more information on digital certificates, refer to

http://www.ietf.org/html.charters/pkix-charter.html and
http://www.ietf.org/rfc/rfc2459.txt .
To determine if there is a digital certificate on a SoundPoint IP, SoundStation IP,

or Polycom VVX phone:
1. Press the Menu key, and then select Status > Platform > Phone.
2. Scroll down to the bottom of screen.
One of three messages will be displayed:
— “Device Certificate: Installed” is displayed if the certificate is available
in flash memory, all the certificate fields are valid (listed above) and
certificate has not expired.
— “Device Certificate: Not Installed” is displayed if the certificate is not
available in flash memory (or the flash memory location where the
device certificate is to be stored is blank).
— “Device Certificate: Invalid” is displayed if the certificate is not valid
(if any of the fields listed above are not correct).
Mutual TLS Authentication

Mutual Transport Layer Security (TLS) authentication is a process in which
both entities in a communications link authenticate each other. In a network
environment, the phone authenticates the server and vice-versa. In this way,
phone users can be assured that they are doing business exclusively with
legitimate entities and servers can be certain that all would-be users are
attempting to gain access for legitimate purposes.
This feature requires that the phone being used has a Polycom
factory-installed device certificate. Refer to the previous section, Digital
Certificates.
Prior to SIP 3.2, and in cases where the phones do not have factory-installed
device certificates, the phone will authenticate to the server as part of the TLS
authentication, but the server cannot cryptographically-authenticate the
phone. This is sometime referred to as Server Authentication or single-sided
Authentication.
Mutual TLS authentication is optional and is initiated by the server. When the
phone acts as a TLS client and the server is configured to require mutual TLS,
the server will request, and then validate the client certificate during the
handshake. If the server is configured to require mutual TLS, a device
certificate and an associated private key must be loaded on the phone.
The digital certificate, stored on the phone, is used by:
• HTTPS device configuration, if the server is configured for Mutual

Authentication
4 - 97
• SIP signaling—when the selected transport protocol is TLS and the server
is configured for Mutual Authentication
• Syslog—when the selected transport protocol is TLS and the server is

configured for Mutual Authentication
• Corporate Directory—when the selected transport protocol is TLS and the

server is configured for Mutual Authentication
Note At this time, the user will not be able to modify or update the digital certificate or the
associated private key stored on the phone during manufacturing.
The Polycom Root CA can be downloaded from http://pki.polycom.com .

The location of the Certificate Revocation List (CRL)—a list of all expired
certificates signed by the Polycom Root CA—is part of the Polycom Root CA
digital certificate. If Mutual TLS is enabled, the Polycom Root CA must be
downloaded onto the HTTPS server.
At this time, the following operating systems/web servers combinations are
supported:
• Microsoft Internet Information Services 6.0 on Microsoft Windows

Server 2003
• Apache v1.3 on Microsoft Windows XP

For more information on using Mutual TLS with Microsoft® Internet
Information Services (IIS) 6.0, refer to “Technical Bulletin 52609: Mutual
Transport Layer Security Provisioning Using Microsoft Internet Information
Services 6.0” at
Configuring SoundPoint IP / SoundStation IP / VVX Phones

Locally
A local phone-based configuration web server is available, unless it is disabled
through sip.cfg. It can be used as the only method of modifying phone
configuration or as a distributed method of augmenting a centralized
provisioning model. For more information, refer to Web Server <httpd/> on
page A-79.
The phone’s local user interface also permits many application settings to be
modified, such as SIP server address, ring type, or regional settings such as
time/date format and language.
4 - 98
Local Web Point your web browser to http://<phoneIPAddress>/.

Server Access Configuration pages are accessible from the menu along the top banner.
The web server will issue an authentication challenge to all pages except for
the home page.
Credentials are (case sensitive):
User Name: Polycom
Password: The administrator password is used for this.
Local Settings Some items in the Settings menu are locked to prevent accidental changes.
Menu Access To unlock these menus, enter the user or administrator passwords.
The administrator password can be used anywhere that the user password is
used. (Polycom recommends that you change the administrative password
from the default value.)
Factory default passwords are:
User password: 123
Administrator password: 456
Passwords:
Administrator Network Configuration

password Line Configuration
required. Call Server Configuration
SSL Security settings
Reset to Default - local configuration, device settings, and file system format
User password Reboot Phone

required.
Changes made through the web server or local user interface are stored
internally as overrides. These overrides take precedence over settings
contained in the configuration obtained from the provisioning server.
If the provisioning server permits uploads, these override setting will be saved
in a file called <Ethernet address>-phone.cfg on the provisioning server as
well as in flash memory.
Warning Local configuration changes will continue to override the provisioning

server-derived configuration until deleted through the Reset Local Config menu
selection or configured using the ‘device set ‘ procedure.
For more information, refer to Modifying Phone’s Configuration Using the

4 - 99
4 - 100
5
Troubleshooting Your SoundPoint IP /
SoundStation IP / VVX Phones
This chapter provides you with some tools and techniques for troubleshooting
SoundPoint IP / SoundStation IP / VVX phones and installations. The phone
can provide feedback in the form of on-screen error messages, status
indicators, and log files for troubleshooting issues.
This chapter includes information on:
• BootROM Error Messages
• SIP Application Error Messages
• Status Menu
• Log Files
• Testing Phone Hardware

This chapter also presents phone issues, likely causes, and corrective actions.
Issues are grouped as follows:
• Power and Startup
• Controls
• Access to Screens and Systems
• Calling
• Displays
• Audio
• Licensable Features
• Upgrading
Review the latest Release Notes for the SIP application for known problems and
possible workarounds. For the latest Release Notes and the latest version of this
Administrator’s Guide, go to Polycom Technical Support at
http://www.polycom.com/support/voice/.
5-1
If a problem is not listed in this chapter nor described in the latest Release Notes,
contact your Certified Polycom Reseller for support.
Error Messages
There are several different error messages that can be displayed on the phone
when it is booting. Some of these errors are fatal, meaning that the phone will
not able to boot until this issue has been resolved, and some are recoverable,
meaning the phone will continue booting after the error, but the configuration
of the phone may not be what you were expecting.
BootROM Error Messages

Most of these errors are also logged on the phone’s boot log, however, if you
are having trouble connecting to the provisioning server, the phone will likely
not be able to upload the boot log for you to examine.
Failed to get boot parameters via DHCP

The phone does not have an IP address and therefore cannot boot. Check that
all cables are connected, the DHCP server is running and that the phone has
not been put into a VLAN which is different from the DHCP server. Check the
DHCP configuration.
Application <file name> is not compatible with this phone!

When the BootROM displays an error like “The application is not compatible”,
it means an application file was downloaded from the provisioning server, but
it cannot be installed on this phone. This issue can usually be resolved by
finding a software image that is compatible with the hardware or the
BootROM being used and installing this on the provisioning server. There are
various different hardware and software dependencies. Refer to the latest
Release Notes for details on the version you are using.
Could not contact boot server, using existing configuration

The phone could not contact the provisioning server, but the causes may be
numerous. It may be cabling issue, it may be related to DHCP configuration,
or it could be a problem with the provisioning server itself. The phone can
recover from this error so long as it previously downloaded a valid application
BootROM image and all of the necessary configuration files.
5-2
Troubleshooting Your SoundPoint IP / SoundStation IP / VVX Phones
Error, application is not present!

There is no application stored in flash memory and the phone cannot boot. A
compatible SIP application must be downloaded into the phone using one of
the supported provisioning protocols. You need to resolve the issue of
connecting to the provisioning server. This error is typically a result of one of
the above errors. This error is fatal, but recoverable. Contact your system
administrator.
Not all configuration files were present on the server

Similarly, a message about configuration files not being present, means that
the phone was able to reach the provisioning server, but that it was not able to
find all the necessary files. So long as the files exist in flash memory, the phone
can boot following this error. The probable cause of this error is a
misconfiguration of the <MACaddress>.cfg file.
Note This error does not occur with BootROM 3.2.2 B or later.
Error loading <file name>

When the required file does not exist in flash memory and cannot be found on
the provisioning server, the “Error loading” message will tell you which file
could not be found. This error only remains on the screen for a few seconds so
you need to watch closely. The phone reboots.
Note This error does not occur with BootROM 3.2.2 B or later.
SIP Application Error Messages
Config file error. Error is <Hex #>

If there is an error in the configuration file, you will not be able to reboot the
phones. You must review the provisioning server configuration, make the
correction, and reapply the configuration file by restarting the phones. This
error also happens when phone does a restart (not a reboot) and finds a newer
version of BootROM or application; this triggers a reboot. Usually this error is
self-recoverable.
Network link is down

Since the SoundPoint IP / SoundStation IP / VVX phones do not have an LED
indicating network LINK status like many networking devices, if a link failure
is detected while the phone is running a message saying “Network link is
down” will be displayed. This message will be shown on the screen whenever
5-3
the phone is not in the menu system and will remain on screen until the link
problem is resolved. Call related functions (for example, soft keys and feature
keys) disabled when the network is down; however the menu works.
Status
When the phone is unable to register with the call control server, the icon
is shown (outline). Once the phone is registered, the icon is
shown (solid). On the SoundStation IP 7000, the icons are and .
On the Polycom VVX 1500, the icons are and .
Flashing Time
If the phone has not been able to contact the SNTP server or if one has not been
configured, the date/time display will flash until this is fixed. If an SNTP is not
available, the data/time display can be turned off so that the flashing display
is not a distraction.
Status Menu
Debugging of single phone may be possible through an examination of the
phone’s status menu. Press Menu, select Status, and then press the Select soft
key.
Under the Platform selection, you can get details on the phone’s serial number
or MAC address, the current IP address, the BootROM version, the application
version, the name of the configuration files in use, and the address of the
In the Network menu, the phone will provide information about TCP/IP
setting, Ethernet port speed, connectivity status of the PC port, and statistics
on packets sent and received since last boot. This would also be a good place
to look and see how long it has been since the phone rebooted. The Call
Statistics screen shows packets sent and received on the last call.
The Lines menu will give you details about the status of each line that has been
configured on the phone.
Finally, the Diagnostics menu offers a series of hardware tests to verify correct
operation of the microphone, speaker, handset, and third party headset, if
present. It will also let you test that each of the keys on the phone is working,
and it will display the function that has been assigned to each of the keys in the
configuration. This is also where you can test the LCD for faulty pixels.
5-4
In addition to the hardware tests, the Diagnostics menu has a series of

real-time graphs for CPU, network and memory utilization that can be helpful
in diagnosing performance issues.
Log Files
SoundPoint IP and SoundStation IP phones will log various events to files
stored in the flash file system and will periodically upload these log files to the
provisioning server. The files are stored in the phone’s home directory or a
user-configurable directory. You can also configure a phone to send log
messages to a syslog server.
There is one log file for the BootROM and one for the application. When a
phone uploads its log files, they are saved on the provisioning server with the
MAC address of the phone prepended to the file name. For example,
0004f200360b-boot.log and 0004f200360b-app.log are the files associated with
MAC address 00f4f200360b. The BootROM log file is uploaded to the
provisioning server after every reboot. The application log file is uploaded
periodically or when the local copy reaches a predetermined size. Refer to
Basic Logging <level/><change/> and <render/> on page A-106.
Both log files can be uploaded on demand using a multiple key combination
described in Multiple Key Combinations on page C-11. The phone uploads
four files, namely, mac-boot.log, app-boot.log, mac-now-boot.log, and
mac-now-app.log. The “now_” logs are uploaded manually unless they are
empty.
The amount of logging that the phone performs can be tuned for the
application to provide more or less detail on specific components of the
phone’s software. For example, if you are troubleshooting a SIP signaling
issue, you are not likely interested in DSP events. Logging levels are adjusted
in the configuration files or via the web interface. You should not modify the
default logging levels unless directed to by Polycom Technical Support.
Inappropriate logging levels can cause performance issues on the phone.
In addition to logging events, the phone can be configured to automatically
execute command-line instructions at specified intervals that output run-time
information such as memory utilization, task status, or network buffer
contents to the log file. These techniques should only be used in consultation
with Polycom Technical Support.
Application Logging Options

Each of the components of the application software is capable of logging
events of different severity. This allows you to capture lower severity events
in one part of the application, while still only getting high severity event for
other components.
5-5
The parameters for log level settings are found in the sip.cfg configuration file.
They are log.level.change.module_name. Log levels range from 1 to 6 (1 for
the most detailed logging, 6 for critical errors only). There are currently 27
different log types that can be adjusted to assist with the investigation of
different problems.
When testing is complete, remember to return all logging levels to the default
value of 4.
There are other logging parameters that you may wish to modify. Changing
these parameters does not have the same impact as changing the logging
levels, but you should still understand how your changes will affect the phone
and the network.
• log.render.level—Sets the lowest level that can be logged (default=1)
• log.render.file.size—Maximum size before log file is uploaded

(default=16 kb)
• log.render.file.upload.period—Frequency of log uploads (default is

172800 seconds = 48 hours)
• log.render.file.upload.append—Controls if log files on the

provisioning server are overwritten or appended, not supported by all
servers
• log.render.file.upload.append.sizeLimit—Controls the maximum

size of log files on the provisioning server (default=512 kb)
• log.render.file.upload.append.limitMode—Controls action to take

when server log reaches max size, actions are stop and delete
Scheduled Logging
Scheduled logging is a powerful tool for anyone who is trying to troubleshoot
an issue with the phone that only occurs after some time in operation.
The output of these instructions is written to the application log, and can be
examined later (for trend data).
The parameters for scheduled logging are found in the sip.cfg configuration
file. They are log.sched.module_name.
5-6
The following figure shows an example of a configuration file and the

resulting log file.
Manual Log Upload

If you want to look at the log files without having to wait for the phone to
upload them (which could take as long as 24 hours or more), initiate an upload
by pressing correct combination of keys on the phone.
For more information, refer to Multiple Key Combinations on page C-11.
When the log files are manually uploaded, the word “now” is inserted into the
name of the file, for example, 0004f200360b-now-boot.log .
5-7
Reading a Boot Log

The following figure shows a portion of a boot log file:
Boot Failure Messages

The following figure shows a number of boot failure messages:
5-8
Reading an Application Log

The following figure shows portions of an application log file:
5-9
Reading a Syslog
The following shows a portion of a syslog log file—the messages look identical
to the normal log with the addition of a timestamp and IP address:
Testing Phone Hardware

To obtain more detailed troubleshooting information, you can access certain
menus on the SoundPoint IP and SoundStation IP phone that test the phone
hardware.
From the diagnostics menu, you can test:
• The phone’s microphones, speaker, handset, and any third-party handset

(if present)
• Keypad mapping—You can verify the function assign to each key.
• Graphic display—You can test the LCD for faulty pixels.
To test the phone hardware:

>> Press Menu, and then select Status > Diagnostics > Test Hardware >
Audio Diagnostics, Keypad Diagnostics, or Display Diagnostics.
5 - 10
Power and Startup

Symptom Problem Corrective Action
There are power issues. The SoundPoint IP / Do one of the following:

SoundStation IP / VVX family SIP • Verify that no lights appear on the unit
phone has no power. when it is powered up.
• Check if the phone is properly plugged
into a functional AC outlet.
• Make sure that the phone isn't
plugged into a plug controlled by a
light switch that is off.
• If plugged into a power strip, try
plugging directly into a wall outlet
instead.
• Try the phone in another room where
the electricity is known to be working
on a particular outlet.
• If using PoE, the power supply voltage
may be too high or too low.
5 - 11
Controls
The dial pad does not work. The dial pad on the SoundPoint Do one of the following:
IP / SoundStation IP / VVX family • Check for a response from other
SIP phone does not respond. feature keys or from the dial pad.
• Place a call to the phone from a known
working telephone. Check for display
updates.
• Press the Menu key followed by
System Status and Server Status to
check if the telephone is correctly
registered to the server.
• Press the Menu key followed by
System Status and Network
Statistics. Scroll down to see if LAN
port shows active or Inactive.
• Check the termination at the switch or
hub end of the network LAN cable.
Ensure that the switch/hub port
connected to the telephone is
operational (if not accessible, contact
your system administrator).
• Before restarting your phone, contact
your system administrator, since this
may allow more detailed
troubleshooting to occur before losing
any current status information.
5 - 12
Access to Screens and Systems

There is no response from The SoundPoint IP / Do one of the following:

feature key presses. SoundStation IP / VVX family SIP • Press the keys more slowly.
phone is not in active state.
• Check to see whether or not the key
has been mapped to a different
function or disabled.
• Make a call to the phone to check for
inbound call display and ringing as
normal. If successful, try to press
feature keys within the call to access
Directory or Buddy Status, for
example.
• Press Menu followed by Status >
Lines to confirm line is actively
registered to the call server.
• Reboot the phone to attempt
re-registration to the call server (refer
to Rebooting the Phone on page
C-11).
The display shows “Network Link The LAN cable is not properly Do one of the following:
is Down”. connected. • Check termination at the switch or hub
(furthest end of the cable from the
phone).
• Check that the switch or hub is
operational (flashing link/status lights)
or contact your system administrator.
• Press Menu followed by Status >
Network. Scroll down to verify that the
LAN is active.
• Ping phone from another machine.
• Reboot the phone to attempt
re-registration to the call server (refer
to Rebooting the Phone on page
C-11).
5 - 13
Calling
There is no dial tone. Power is not correctly applied to Do one of the following:
the SoundPoint IP family SIP • Check that the display is illuminated.
phone.
• Make sure the LAN cable is inserted
properly at the rear of the phone (try
unplugging and re-inserting the
cable).
• If using in-line powering, have your
system administrator check that the
switch is supplying power to the
phone.
Dial tone is not present on one of Do one of the following:

audio paths. • Switch between Handset, Headset (if
present) or Hands-Free
Speakerphone to see if dial tone is
present on another paths.
• If dial tone exists on another path,
connect a different handset or
headset to isolate the problem.
• Check configuration for gain levels.
The phone is not registered. Contact your system administrator.
The phone does not ring. Ring setting or volume is low. Do one of the following:
• Adjust the ringing level from the front
panel using the volume up/down keys.
• Check same status of handset,
headset (if connected) and through
the Hands-Free Speakerphone.
Outbound or inbound calling is Do one of the following:

unsuccessful. • Place a call to the phone under
investigation. Check that the display
indicates incoming call information.
• Lift the handset. Ensure dial tone is
present and place a call to another
extension or number. Check that the
display changes in response.
The line icon shows an The phone line is unregistered. Contact your system administrator.
unregistered line icon.
5 - 14
Displays
There is no display. Power is not correctly applied to Do one of the following:

The display is incorrect. the SoundPoint IP family SIP • Check that the display is illuminated.
phone.
The display has bad contrast. • Make sure the LAN cable is inserted
properly at the rear of the phone (try
unplugging and re-inserting the
cable).
• If using in-line powering, have your
system administrator check that the
switch is supplying power to the
phone.
• Use the screen capture feature. Refer
to Capturing Phone’s Current Screen
on page C-30.
The contrast needs adjustment. Do one of the following:

• Refer to the appropriate SoundPoint
IP / SoundStation IP / VVX SIP phone
User Guide.
• Reboot the phone to obtain a default
level of contrast (refer to Rebooting
the Phone on page C-11).
on page C-30.
Outbound or inbound calling is Do one of the following:

unsuccessful. • Place a call to the phone under
investigation. Check that the display
indicates incoming call information.
• Lift the handset. Ensure dial tone is
present and place a call to another
extension or number. Check that the
display changes in response.
on page C-30.
The display is flickering. Certain type of older fluorescent Do one of the following:
lighting causes the display to • Move the SoundPoint IP /
appear to flicker. SoundStation IP / VVX SIP phone
away from the lights.
• Replace the lights.
on page C-30.
5 - 15
Audio
There is no audio on the The connections are not correct. Do one of the following:
headset. • Ensure the headset is plugged into the
jack marked Headset at the rear of the
phone.
• Ensure the headset amplifier (if
present) is turned on and/or the
volume is correctly adjusted).
There are audio and echo issues Possible issues include: Refer to “Technical Bulletin 16249:
on the headset. • Echo on external calls Troubleshooting Audio and Echo Issues
through a gateway. on SoundPoint® IP Phones” at
http://www.polycom.com/usa/en/support/v
• Internal calls (no gateway),
oice/soundpoint_ip/VoIP_Technical_Bullet
handsfree echo.
ins_pub.html .
• Internal calls (no gateway),
handset to handset echo.
Licensable Features
A user is trying to access one of The license is not installed on the Do the following:
the following features, but it is not phone or it has expired. • Press the Menu key, then select
available on their phone: Status > Licenses.
• Corporate Directory • Using the arrow keys, verify that the
• Recording and Playback of feature in question has a valid license.
Audio Calls If no licenses are installed, the “No
• Managing Conferences license installed.” message appears.
• Voice Quality Monitoring
• H.323
5 - 16
Upgrading
SoundPoint IP 300, 301, 500, New features are not supported The attempt to load the new application
501, 600, and/or 601 and/or on the SoundPoint IP 300, 301, will fail since there is no
SoundStation IP 4000 behave 500, 501, 600, and 601 and 300/301/500/501/600/601/4000 image
incorrectly or do not display new SoundStation IP 4000 and the contained within the sip.ld file, so the
features. configuration files have not been phone will continue on and run the current
correctly modified. These phones version of application that it has in
will not ‘understand’ the new memory. It will however use the new
configuration parameters, and configuration files. Refer to Supporting
will attempt to load the new SoundPoint IP 300, 301, 500, 501, 600
application. and 601 and SoundStation IP 4000
Phones on page 3-23.
5 - 17
5 - 18
A
Configuration Files
This appendix provides detailed descriptions of certain configuration files

used by the Session Initiation Protocol (SIP) application. It is a reference for all
parameters that are configurable when using the centralized provisioning
installation model.
This appendix contains information on:
• Master Configuration Files (MAC-address.cfg or 000000000000.cfg)
• Application Configuration (sip.cfg)
• Per-Phone Configuration (phone1.cfg)
• Flash Parameter Configuration

The application configuration files dictate the behavior of the phone once it is
running the executable specified in the master configuration file.
Warning Configuration files should only be modified by a knowledgeable system

administrator. Applying incorrect parameters may render the phone unusable. The
configuration files which accompany a specific release of the SIP software must be
used together with that software. Failure to do this may render the phone unusable.
Note In the tables in the subsequent sections, “Null” should be interpreted as the empty
string, that is, attributeName=“” when the file is viewed in an XML editor.
To enter special characters in a configuration file, enter the appropriate sequence
using an XML editor:
• & as &
• ” as "
• ’ as '
• < as <
• > as >
Note The various .hd. parameters in sip.cfg (such as voice.aec.hd.enable,

voice.ns.hd.enable, and voice.agc.hd.enable) are headset parameters. They
are not connected to high definition or HD voice.
A-1
Note You can make changes to the configuration files through the web interface to the
phone. Using your chosen browser, enter the phone’s IP address as the browser
address. For more information, refer to Modifying Phone’s Configuration Using the
Changes made through the web interface are written to the override file (highest
priority). These changes remain active until Reset Local Config is performed.
Master Configuration Files

The master configuration files can be one of:
• Specified master configuration file—The master configuration file can be

explicitly specified in the provisioning server address, for example,
http://usr:pwd@server/dir/example1.cfg. The filename must end with
.cfg and be at least five characters long. If this file cannot be downloaded,
the phone will search for the per-phone master configuration file
(described next).
• Per-phone master configuration file—If per-phone customization is

required, the file should be named <Ethernet address>.cfg, where
Ethernet address is the MAC address of the phone in question. For A-F
hexadecimal digits, use upper or lower case, for example,
0004f200106c.cfg. The Ethernet address can be viewed using the About
soft key during the auto-restart countdown of the BootROM or through
the Menu > Status > Platform > Phone menu in the application. It is also
printed on a label on the back of the phone. If this file cannot be
downloaded, the phone will search for the default master configuration
file (described next).
• Default master configuration file—For systems in which the configuration

is identical for all phones (no per-phone <Ethernet address>.cfg files), the
default master configuration file may be used to set the configuration for
all phones. The file named 000000000000.cfg (<12 zeros>.cfg) is the default
master configuration file and it is recommended that one be present on the
provisioning server. If a phone does not find its own <Ethernet
address>.cfg file, it will use this one, and establish a baseline
configuration. This file is part of the standard Polycom distribution of
configuration files. It should be used as the template for the <Ethernet
address>.cfg files.
The default master configuration file, 000000000000.cfg, for SIP 3.2.2 is
shown below:
<?xml version=”1.0” standalone=”yes”?>




A-2
Configuration Files


<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg,
sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY=""
CONTACTS_DIRECTORY="" LICENSE_DIRECTORY="">
<APPLICATION_SPIP300 APP_FILE_PATH_SPIP300="sip_212.ld"
CONFIG_FILES_SPIP300="phone1_212.cfg, sip_212.cfg"/>
<APPLICATION_SSIP4000 APP_FILE_PATH_SSIP4000="sip_313.ld"
CONFIG_FILES_SSIP4000="phone1_313.cfg, sip_313.cfg"/>
</APPLICATION>
Master configuration files contain the following XML attributes:
• APP_FILE_PATH—The path name of the application executable. It can

have a maximum length of 255 characters. This can be a URL with its own
protocol, user name and password, for example
http://usr:pwd@server/dir/sip.ld.
• CONFIG_FILES—A comma-separated list of configuration files. Each file

name has a maximum length of 255 characters and the list of file names has
a maximum length of 2047 characters, including commas and white space.
Each configuration file can be specified as a URL with its own protocol,
user name and password, for example
ftp://usr:pwd@server/dir/phone2034.cfg.
• MISC_FILES—A comma-separated list of other required files. Dictionary

resource files listed here will be stored in the phone's flash file system. So
if the phone reboots at a time when the provisioning server is unavailable,
it will still be able to load the preferred language.
• LOG_FILE_DIRECTORY—An alternative directory to use for log files if

required. A URL can also be specified. This is blank by default.
• CONTACTS_DIRECTORY—An alternative directory to use for user

directory files if required. A URL can also be specified. This is blank by
default.
• OVERRIDES_DIRECTORY—An alternative directory to use for

configuration overrides files if required. A URL can also be specified. This
is blank by default.
A-3
• LICENSE_DIRECTORY—An alternative directory to use for license files if

required. A URL can also be specified. This is blank by default.
Warning The order of the configuration files listed in CONFIG_FILES is significant:

• The files are processed in the order listed (left to right).
• The same parameters may be included in more than one file.
• The parameter found first in the list of files will be the one that is effective.
This provides a convenient means of overriding the behavior of one or more phones
without changing the baseline configuration files for an entire system.
If you have a requirement for different application loads on different phones

on the same provisioning server, you can create a variable in the master
configuration file that is replaced by the MAC address of each phone when it
reboots. An example is shown below:
refer to the -->


<!--
<APPLICATION APP_FILE_PATH=”sip.ld” CONFIG_FILES=”phone1.cfg,
sip.cfg” MISC_FILES=”” LOG_FILE_DIRECTORY=”” OVERRIDES_DIRECTORY=””
CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=”” />
<APPLICATION APP_FILE_PATH_SPIP300=”SPIP300.sip.ld”
CONFIG_FILES_SPIP300=”phone1_SPIP300.cfg, sip_SPIP300.cfg” />
<APPLICATION APP_FILE_PATH_SPIP500=”SPIP500.sip.ld”
CONFIG_FILES_SPIP500=”phone1_SPIP500.cfg, sip_SPIP500.cfg” />
For more information:
• Refer to “Technical Bulletin 35311: Supporting SoundPoint IP 300, 301,

500, 501, 600, and 601 and SoundStation IP 4000 Phones with SIP 2.2.0 or
SIP 3.2.0 and Later Releases“ at
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoI
P_Technical_Bulletins_pub.html .
• Refer to “Technical Bulletin 35361: Overriding Parameters in Master

Configuration File on SoundPoint IP Phones“ at
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoI
P_Technical_Bulletins_pub.html.
Application Configuration
The configuration file sip.cfg contains SIP protocol and core configuration
settings that would typically apply to an entire installation and must be set
before the phones will be operational, unless changed through the local web
server interface or local menu settings on the phone. These settings include the
local port used for SIP signaling, the address and ports of a cluster of SIP
application servers, voice codecs, gains, and tones, and other parameters.
A-5
For more information, refer to the“Configuration File Management on SoundPoint
These parameters include:
• Protocol <voIpProt/>
• Dial Plan <dialplan/>
• Localization <lcl/>
• User Preferences <up/>
• Tones <tones/>
• Sampled Audio for Sound Effects <saf/>
• Sound Effects <se/>
• Voice Settings <voice/>
• Video Settings <video/>
• Basic TCP/IP <TCP_IP/>
• Web Server <httpd/>
• Call Handling Configuration <call/>
• Directory <dir/>
• Presence <pres/>
• Fonts <font/>
• Keys <key/>
• Backgrounds <bg/>
• Bitmaps <bitmap/>
• Indicators <ind/>
• Event Logging <log/>
• Security <sec/>
• License <license/>
• Provisioning <prov/>
• RAM Disk <ramdisk/>
A-6
Configuration Files
• Request <request/>
• Feature <feature/>
• Resource <res/>
• Microbrowser <mb/>
• Applications <apps/>
• Peer Networking <pnet/>
• DNS Cache <dns/>
• Soft Keys <softkey/>
• LCD Power Saving <powerSaving/>
Protocol <voIpProt/>
This attribute includes:
• Server <server/>
• SIP <SIP/>
• H.323 <H323/>
A-7
Server <server/>
This configuration attribute is defined as follows:
Permitted
Attribute Values Default Interpretation
voIpProt.server.dhcp.available 0 or 1 0 If set to 1, check with the DHCP server for

SIP server IP address. If set to 0, do not
check with DHCP server.
voIpProt.server.dhcp.option 128 to 255 Null Option to request from the DHCP server if
voIpProt.server.dhcp.available = 1.
There is no default value for this parameter, it
must be filled in with a valid value.
Note: If the reg.x.server.y.address parameter
in Registration <reg/> on page A-134 is
non-Null, it takes precedence even if the
DHCP server is available.
voIpProt.server.dhcp.type 0 or 1 Null If set to 0, IP request address.

If set to 1, request string.
Type to request from the DHCP server if
voIpProt.server.dhcp.available = 1.
There is no default value for this parameter, it
must be filled in with a valid value.
A-8
Configuration Files
Permitted
voIpProt.server.x.address dotted-deci Null IP address or host name and port of a SIP

mal IP server that accepts registrations. Multiple
address or servers can be listed starting with x=1, 2, ...
host name for fault tolerance.
If port is 0 or Null:
voIpProt.server.x.port 0, Null, 1 to Null
If voIpProt.server.x.address is a
65535
hostname and
voIpProt.server.x.transport is set to
DNSnaptr, do NAPTR then SRV lookups.
If voIpProt.server.x.transport is set to
TCPpreferred or UDPOnly, then use 5060
and don’t advertise the port number in
signaling.
If voIpProt.server.x.address is an IP
address, there is no DNS lookup and 5060 is
used for the port but it is not advertised in
signaling.
If port is 1 to 65535:
This value is used and it is advertised in
signaling.
Note: If the reg.x.server.y.address parameter
in Registration <reg/> on page A-134 is
non-Null, all of the reg.x.server.y.xxx
parameters will override the voIpProt.server
parameters.
Note: The H.323 gatekeeper RAS signaling
uses UDP, while the H.225/245 signaling
uses TCP.
voIpProt.server.x.transport DNSnaptr or DNSnapt If set to Null or DNSnaptr:

TCPpreferre r If voIpProt.server.x.address is a
d or hostname and voIpProt.server.x.port is 0 or
UDPOnly or Null, do NAPTR then SRV look-ups to try to
TLS or discover the transport, ports and servers, as
TCPOnly per RFC 3263. If
voIpProt.server.x.address is an IP
address, or a port is given, then UDP is used.
If set to TCPpreferred:
TCP is the preferred transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport fails. Leave port field
empty (will default to 5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
A-9
Permitted
voIpProt.server.x.expires positive 3600 The phone’s requested registration period in

integer, seconds.
minimum 10 Note: The period negotiated with the server
may be different. The phone will attempt to
re-register at the beginning of the overlap
period. For example, if “expires”=300 and
“overlap”=5, the phone will re-register after
295 seconds (300-5).
voIpProt.server.x.expires.overlap positive 60 The number of seconds before the expiration

integer, time returned by server x at which the phone
minimum 5, should try to re-register. The phone will try to
maximum re-register at half the expiration time returned
65535 by the server if that value is less than the
configured overlap value.
voIpProt.server.x.register 0 or 1 1 If set to 0, calls can be routed to an outbound

proxy without registration. Refer to
reg.x.server.y.register in Registration
<reg/>on page A-134.
For more information, refer to “Technical
Bulletin 5844: SIP Server Fallback
Enhancements on SoundPoint IP Phones“ at
http://www.polycom.com/usa/en/support/voic
e/soundpoint_ip/VoIP_Technical_Bulletins_p
ub.html
voIpProt.server.x.retryTimeOut Null or 0 If set to 0 or Null, use standard RFC 3261

non-negativ signaling retry behavior. Otherwise
e integer retryTimeOut determines how often retries
will be sent.
Units = milliSeconds. (Finest resolution =
100ms).
voIpProt.server.x.retryMaxCount Null or 3 If set to 0 or Null, 3 is used. retryMaxCount

non-negativ retries will be attempted before moving on to
e integer the next available server.
voIpProt.server.x.expires.lineSeize positive 30 Requested line-seize subscription period.

integer,
minimum 10
voIpProt.server.x.lcs 0 or 1 0 This attribute overrides the

voIpProt.SIP.lcs .
If set to 1, the proprietary “epid” parameter is
added to the From field of all requests to
support Microsoft Live Communications
Server.
A - 10
Configuration Files
Permitted
voIpProt.server.H323.x.address dotted-deci Null Address of the H.323 gatekeeper.

mal IP Note: Only one H.323 gatekeeper per phone
address or is supported; if more than one is configured,
host name only the first is used.
voIpProt.server.H323.x.port 0, Null, 1 to Null Port to be used for H.323 signaling.

65535 If set to Null, 1719 (H.323 RAS signaling) is
used.
Note: The H.323 gatekeeper RAS signaling
uses UDP, while the H.225/245 signaling
uses TCP.
voIpProt.server.H323.x.expires positive Null Desired registration period.

integer
SIP <SIP/>
Permitted
voIpProt.SIP.enable 0 or 1 1 Flag to determine whether or not the SIP

protocol is used for call routing, dial plan,
DTMF, and URL dialing.
If set to 1, the SIP protocol is used.
voIpProt.SIP.local.port 0 to 65535 5060 Local port to be used for SIP signaling.

Local port for sending and receiving SIP
signaling packets.
If set to 0 or Null, 5060 is used for the local
port but it is not advertised in the SIP
signaling.
If set to some other value, that value is used
for the local port and it is advertised in the
SIP signaling.
voIpProt.SIP.useContactInReferTo 0 or 1 0 If set to 0, the “To URI” is used in the REFER.

If set to 1, the “Contact URI” is used in the
REFER.
voIpProt.SIP.useRFC2543hold 0 or 1 0 If set to 1, use SDP media direction attributes

(such as a=sendonly) per RFC 3264 when
initiating a call, otherwise use the obsolete
c=0.0.0.0 RFC2543 technique. In either case,
the phone processes incoming hold signaling
in either format.
Note: voIpProt.SIP.useRFC2543hold is
effective only when the call is initiated.
A - 11
Permitted
voIpProt.SIP.useSendonlyHold 0 or 1 1 If set to 1, the phone will send a reinvite with

a stream mode attribute of “sendonly” when a
call is put on hold. This is the same as the
previous behavior.
If set to 0, the phone will send a reinvite with
a stream mode attribute of “inactive” when a
call is put on hold.
NOTE: The phone will ignore the value of this
parameter if set to 1 when the parameter
voIpProt.SIP.useRFC2543hold is also set
to 1 (default is 0).
voIpProt.SIP.lcs 0 or 1 0 If set to 1, the proprietary “epid” parameter is

added to the From field of all requests to
support Microsoft Live Communications
Server.
voIpProt.SIP.ms-forking 0 or 1 0 If set to 0, support for MS-forking is disabled.

If set to 1, support for MS-forking is enabled
and the phone will reject all Instant Message
INVITEs. This parameter is relevant for
Microsoft Live Communications Server
server installations.
Note that if any end point registered to the
same account has MS-forking disabled, all
other end points default back to non-forking
mode. Windows Messenger does not use
MS-forking so be aware of this behavior if
one of the end points is Windows Messenger.
voIpProt.SIP.sendCompactHdrs 0 or 1 0 If set to 0, SIP header names generated by

the phone use the long form, for example
‘From’.
If set to 1, SIP header names generated by
the phone use the short form, for example ‘f’.
voIpProt.SIP.keepalive. 0 or 1 0 If set to 1, the session timer will be enabled.

sessionTimers If set to 0, the session timer will be disabled,
and the phone will not declare “timer” in
“Support” header in INVITE. The phone will
still respond to a re-INVITE or UPDATE. The
phone will not try to re-INVITE or do UPDATE
even if remote end point asks for it.
voIpProt.SIP.requestURI.E164. 0 or 1 0 If set to 1, ‘+’ global prefix is added to E.164

addGlobalPrefix user parts in sip: URIs:.
A - 12
Configuration Files
Permitted
voIpProt.SIP. 0 or 1 1 If set to 1, a transfer can be completed during

allowTransferOnProceeding the proceeding state of a consultation call.
If set to 0, a transfer is not allowed during the
proceeding state of a consultation call.
If set to Null, the default value is used.
voIpProt.SIP.pingInterval 0 to 3600 0 The number in seconds to send "PING"

message. This feature is disabled by default.
voIpProt.SIP.useContactInReferTo 0 or 1 0 If set to 1, the Contact URI is used.

If set to 0, the TO URI is used (previous
behavior).
voIpProt.SIP.serverFeatureControl.cf 0 or 1 Null If set to 1, server-based call forwarding is

enabled. The call server has control of call
forwarding.
If set to 0 or Null, server-based call
forwarding is not enabled. This is the old
behavior.
voIpProt.SIP.serverFeatureControl.loc 0 or 1 Null If set to 0 and

alProcessing.cf voIpProt.SIP.serverFeatureControl.cf=
"1", the phone will not perform local Call
Forward behavior.
If set to 1 or Null, the phone will perform local
Call Forward behavior on all calls received.
voIpProt.SIP.serverFeatureControl. 0 or 1 Null If set to 1, server-based DND is enabled. The

dnd call server has control of DND.
If set to 0 or Null, server-based DND is not
enabled. This is the old behavior.
voIpProt.SIP.serverFeatureControl. If set to 0 and

localProcessing.dnd voIpProt.SIP.serverFeatureControl.dnd
="1", the phone will not perform local DND
call behavior.
If set to 1 or Null, the phone will perform local
DND call behavior on all calls received.
A - 13
Permitted
voIpProt.SIP.authOptimizedInFailover 0,1 0 If set to 1, when failover occurs, the first new

SIP request is sent to the server that sent the
proxy authentication request.
If set to 0, when failover occurs, the first new
SIP request is sent to the server with the
highest priority in the server list.
If reg.x.auth.optimizedInFailover set to
Null, this attribute is checked.
If
voIpProt.SIP.authOptimizedInFailover
is Null, then this feature is disabled.
If both attributes are set, the value of
reg.x.auth.optimizedInFailover takes
precedence.
voIpProt.SIP.csta 0 or 1 0 If set to 1, uaCSTA is enabled.
voIpProt.SIP.strictLineSeize 0 or 1 Null If set to 1, forces the phone to wait for 200

OK response when receiving a TRYING
notify.
If set to 0 or Null, this is old behavior.
voIpProt.SIP.strictUserValidation 0 or 1 Null If set to 1, forces the phone to match user

portion of signaling exactly.
If set to 0 or Null, phone will use first
registration if the user part does not match
any registration.
voIpProt.SIP.lineSeize.retries 3 to 10 10 Controls the number of times the phone will

retry a notify when attempting to seize a line
(BLA).
voIpProt.SIP.header.diversion.enable 0 or 1 0 If set to 1, the diversion header is displayed if

received.
If set to 0 or Null, the diversion header is not
displayed.
voIpProt.SIP.header.diversion.list. 0 or 1 1 If set to 1 or Null, the first diversion header is

useFirst displayed.
If set to 0, the last diversion header is
displayed.
A - 14
Configuration Files
Permitted
voIpProt.SIP.header.warning.codes. comma Null A list of accepted warning codes.

accept separated If set to Null, all codes are accepted. Only
list codes between 300 and 399 are supported.
For example, if you want to accept only
codes 325 to 330:
voIpProt.SIP.header.warning.codes.acc
ept = 325,326,327,328,329,330
Text will be shown in the appropriate
language. For more information, refer to
lcl.ml.lang.tags.x in Multilingual <ml/>
on page A-28.
voIpProt.SIP.header.warning.enable 0 or 1 0 If set to 1, the warning header is displayed if

received.
If set to 0 or Null, the warning header is not
displayed.
voIpProt.SIP.acd.signalingMethod 0 or 1 0 If set to 0 or Null, the ‘SIP-B’ signaling is

supported. (This is the older ACD
functionality.)
If set to 1, the feature synchronization
signaling is supported. (This is the new ACD
functionality.)
voIpProt.SIP.tcpFastFailover 0 or 1 Null If set to 1, failover occurs based on the

values of reg.x.server.y.retryMaxCount
voIpProt.server.x.retryTimeOut.
If set to 0, this is old behavior.
If reg.x.tcpFastFailover is Null, this
attribute is checked.
If voIpProt.SIP.tcpFastFailover is Null,
then this feature is disabled.
If both attributes are set, the value of
reg.x.tcpFastFailover takes precedence.
A - 15
Permitted
voIpProt.SIP.strictReplacesHeader 0 or 1 Null This parameter applies only to directed call

pick-up attempts initiated against monitored
BLF resources.
If set to 1 or Null, the phone requires
call-id,to-tag, and from-tag to perform a
directed call-pickup when
call.directedCallPickupMethod is configured
as "native".
If set to 0, all that is required to perform the
directed call pick-up is a call-id.
voIpProt.SIP.use486forReject 0 or 1 0 If set to1 and the phone is indicating a ringing

inbound call appearance, phone will transmit
a 486 response to the received INVITE when
the Reject soft key is pressed.
If set to 0, no 486 response is transmitted.
voIpProt.SIP.dtmfViaSignaling. 0 or 1 0 If set to 1, DTMF digit information is sent in

rtc2976 RFC2976 SIP INFO packets during a call.
If set to 0 or Null, no DTMF digit information
is sent.
This attribute also includes:
• SDP <SDP/>
• Outbound Proxy <outboundProxy/>
• Alert Information <alertInfo/>
• Request Validation <requestValidation/>
• Special Events <specialEvent/>
• Conference Setup <conference/>
• Dialog <dialog/>
• Connection Reuse<dialog/>
• Music on Hold <musicOnHold/>
• Compliance <compliance/>
A - 16
Configuration Files
SDP <SDP/>
Permitted
voIpProt.SDP. 0 or 1 Null If set to 1, the phone transmits and receives

useLegacyPayloadTypeNegotiation RTP using the payload type identified by the
first codec listed in the SDP of the codec
negotiation answer.
If set to 0 or Null, RFC 3264 is followed for
transmit and receive RTP payload type
values.
voIpProt.SDP.answer. 0 or 1 0 If set to 1, the phones uses its own

useLocalPreferences preference list when deciding which codec to
use rather than the preference list in the offer.
If set to 0, it is disabled.
Note: If a H.323 call from a Polycom VVX
1500 selects a lower-quality codec (H.261)
but the called device also support H.264, this
parameter should be enabled to resolve the
situation.
voIpProt.SDP.iLBC.13_33kbps. 0 or 1 Null If set to 1 or Null, the phone should include

includeMode the mode=30 FMTP attribute in SDP offers:
• If voice.codecPref.iLBC.13_33kbps is
set and
voice.codecPref.iLBC.15_2kbps is
Null.
• If voice.codecPref.iLBC.13_33kbps
and voice.codecPref.iLBC.15_2kbps
are both set, but iLBC 13.33 kbps codec
is set to a higher preference.
If set to 0, the phone should not include the
mode=30 FTMP attribute in SDP offers even
if iLBC 13.33 kbps codec is being advertised.
Refer to Codec Preferences <codecPref/> on
page A-45.
voIpProt.SDP.early.answerOrOffer 0 or 1 Null If set to 1, an SDP offer or answer is

generated in a provisional reliable response
and PRACK request and response.
If set to 0, an SDP offer or answer is not
generated.
Note: An SDP offer or answer is not
generated if the user (reg.x) is configured for
the Music On Hold. Refer to Music on Hold
<musicOnHold/> on page A-21.
A - 17
Outbound Proxy <outboundProxy/>

Permitted
voIpProt.SIP.outboundProxy.address dotted-deci Null IP address or host name and port of a SIP

mal IP server to which the phone shall send all
address or requests.
host name
voIpProt.SIP.outboundProxy.port 0 to 65535 5060
voIpProt.SIP.outboundProxy. DNSnaptr or DNSnapt If set to Null or DNSnaptr:

transport TCPpreferre r If voIpProt.SIP.outboundProxy.address is a
d or hostname and
UDPOnly or voIpProt.SIP.outboundProxy.port is 0 or
TLS or Null, do NAPTR then SRV look-ups to try to
TCPOnly discover the transport, ports and servers, as
per RFC 3263. If
voIpProt.SIP.outboundProxy.address is
an IP address, or a port is given, then UDP is
used.
TCP fails.
If set to UDPOnly:
If set to TLS:
If set to TCPOnly:
NOTE: TLS is not supported on SoundPoint
IP 300 and 500 phones.
A - 18
Configuration Files
Alert Information <alertInfo/>

Permitted
voIpProt.SIP.alertInfo.x.value string to Null Alert-Info fields from INVITE requests will be

compare compared against as many of these
against the parameters as are specified (x=1, 2, ..., N)
value of and if a match is found, the behavior
Alert-Info described in the corresponding ring class
headers in (refer to Ring type <rt/> on page A-43) will be
INVITE applied.
requests
voIpProt.SIP.alertInfo.x.class positive Null

integer
Request Validation <requestValidation/>

Permitted
voIpProt.SIP.requestValidation.x. One of: Null Sets the name of the method for which
request “INVITE”, “ACK” validation will be applied.
, “BYE” WARNING: Intensive request validation
“REGISTER”, may have a negative performance impact
“CANCEL”, due to the additional signaling required in
“OPTIONS”, some cases, therefore, use it wisely.
“INFO”,
“MESSAGE”,
“SUBSCRIBE”
“NOTIFY”,
“REFER”,
“PRACK”, or
“UPDATE”
voIpProt.SIP.requestValidation.x. Null or Null If Null, no validation is done. Otherwise this

method one of: “source”, sets the type of validation performed for the
“digest” or request:
“both”/”all” source: ensure request is received from an
IP address of a server belonging to the set
of target registration servers;
digest: challenge requests with digest
authentication using the local credentials
for the associated registration (line);
both or all: apply both of the above methods
A - 19
Permitted
voIpProt.SIP.requestValidation.x. A valid string Null Determines which events specified with the
request.y.event Event header should be validated; only
applicable when
voIpProt.SIP.requestValidation.x.re
quest is set to “SUBSCRIBE” or “NOTIFY”.
If set to Null, all events will be validated.
voIpProt.SIP.requestValidation. A valid string Polycom Determines string used for Realm.

digest.realm SPIP
Special Events <specialEvent/>

Permitted
voIpProt.SIP.specialEvent.lineSeize. 0 or 1 1 If set to 1, process a 200 OK response for a

nonStandard line-seize event SUBSCRIBE as though a
line-seize NOTIFY with Subscription State:
active header had been received, this speeds
up processing.
voIpProt.SIP.specialEvent. 0 or 1 0 If set to 1, always reboot when a NOTIFY

checkSync.alwaysReboot message is received from the server with
event equal to check-sync.
If set to 0, only reboot if any of the files listed
in <MAC-address>.cfg have changed on the
FTP server when a NOTIFY message is
received from the server with event equal to
check-sync.
Conference Setup <conference/>

Permitted
voIpProt.SIP.conference.address ASCII string Null If Null, conferences are set up on the phone
up to 128 locally.
characters If set to some value, conferences are set up
long by the server using the conferencing agent
specified by this address. The acceptable
values depend on the conferencing server
implementation policy.
A - 20
Configuration Files
Dialog <dialog/>
Permitted
voIpProt.SIP.dialog.useSDP 0 or 1 0 If set to 0, new dialog event package draft is

used (no SDP in dialog body).
If set to 1, for backwards compatibility, use
this setting to send SDP in dialog body.
voIpProt.SIP.dialog.usePvalue 0 or 1 0 If set to 0, phone uses "pval" field name in

Dialog. This obeys the
draft-ietf-sipping-dialog-package-06.txt draft.
If set to 1, phone uses a field name of
"pvalue".
Connection Reuse<dialog/>
Permitted
voIpProt.SIP.connectionReuse. 0 or 1 0 If set to 0, this is the old behavior.

useAlias If set to 1, phone uses the connection reuse
draft which introduces "alias".
Music on Hold <musicOnHold/>

Permitted
voIpProt.SIP.musicOnHold.uri string Null A URI that provides the media stream to play
for the remote party on hold.
If reg.x.musicOnHold is set to Null, this
attribute is checked.
Note: The SIP URI parameter transport is
supported when configured with the values of
UDP, TCP, or TLS.
A - 21
Compliance <compliance/>
Permitted
voIpProt.SIP.compliance.RFC3261. 0 or 1 Null If set to 1, validation of the SIP header

validate.contentLanguage content language is enabled.
If set to 0 or Null, validation is disabled.
voIpProt.SIP.compliance.RFC3261. 0 or 1 Null If set to 1 or Null, validation of the SIP header

validate.uriScheme URI scheme is enabled.
If set to 0, validation is disabled.
H.323 <H323/>
Note At this time, this attribute is used with the Polycom VVX 1500 phone only.
Permitted
voIpProt.H323.enable 0 or 1 0 Flag to determine whether or not the H.323

protocol is used for call routing, dial plan,
DTMF, and URL dialing.
If set to 1, the H.323 protocol is used.
voIpProt.H323.local.port 0 to 65535 1720 Local port to be used for H.323 signaling.

Local port for sending and receiving H.323
signaling packets.
If set to 0 or Null, 1720 is used for the local
port but it is not advertised in the H.323
signaling.
If set to some other value, that value is used
for the local port and it is advertised in the
H.323 signaling.
voIpProt.H323. 0 or 1 0 If set to 1, the phone will attempt to discover

autoGatekeeperDiscovery an H.323 gatekeeper address via the
standard multicast technique, provided that a
statically configured gatekeeper address is
not available.
If set to 0, the phone will no send out any
gatekeeper discovery messages.
voIpProt.H323.dtmfViaSignaling. 0 or 1 1 If set to 1, the phone will use the H.323

enabled signaling channel for DTMF key press
transmission.
A - 22
Configuration Files
Permitted
voIpProt.H323.dtmfViaSignaling. 0 or 1 1 If set to 1, the phone will support H.245

H245alphanumericMode signaling channel alphanumeric mode DTMF
transmission.
Note: If both alphanumeric and signal mode
are enabled, the phone will give preference
to sending DTMF in alphanumeric mode
where there is the possibility of sending in
both modes.
voIpProt.H323.dtmfViaSignaling. 0 or 1 1 If set to 1, the phone will support H.245

H245signalMode signaling channel signal mode DTMF
transmission.
Dial Plan <dialplan/>
Note The dial plan is not applied against Placed Call List, VoiceMail, last call return,
remote control dialed numbers, and on-hook dialing.
Permitted
dialplan.applyToCallListDial 0 or 1 0 This attribute covers dialing from Received

Call List and Missed Call List including dialing
from Edit or Info sub- menus.
If set to 0, the digit map replacement
operations are not applied against the dialed
number.
if set to 1, the digit map replacement
operations are applied against the dialed
number.
dialplan.applyToDirectoryDial 0 or 1 0 This attribute covers dialing from Directory as

well as Speed Dial List.
Value interpretation is the same as for
dialplan.applyToCallListDial.
Note: An Auto Call Contact number is
considered a dial from directory.
dialplan.applyToUserDial 0 or 1 1 This attribute covers the case when the user

presses the Dial soft key to send dialed
number when in idle state display.
A - 23
Permitted
dialplan.applyToUserSend 0 or 1 1 This attribute covers the case when the user

presses the Send soft key to send the dialed
number.
dialplan.impossibleMatchHandling 0, 1 or 2 0 Affects digits entered while in dial mode. For

example, the digits are affected after a user
has picked up the handset, headset, or
pressed the dial key, and not when hot
dialing, contact dialing, or call list dialing.
If set to 0, the digits entered up to and
including the point where an impossible
match occurred are sent to the server
immediately.
If set to 1, give reorder tone.
If set to 2, allow user to accumulate digits and
dispatch call manually with the Send soft key.
dialplan.removeEndOfDial 0 or 1 1 If set to 1, strip trailing # digit from digits sent

out.
dialplan.applyToTelUriDial 0 or 1 1 A flag to determine if the dial plan applies to

uses of the tel:// URI.
If set to 1 or Null, the dial plan applies.
If set to 0, the dial plan does not apply.
dialplan.applyToRemoteDialing 0 or 1 0 A flag to determine if the dial plan applies to

for calls made through the Polycom HDX or
SoundStructure systems.
If set to 1, the dial plan applies.
If set to 0 or Null, the dial plan does not apply.
This attributes also includes:
• Digit Map <digitmap/>
• Routing <routing/>
Digit Map <digitmap/>

A digit map is defined either by a “string” or by a list of strings. Each string in
the list is an alternative numbering scheme, specified either as a set of digits or
timers, or as an expression over which the gateway will attempt to find a
shortest possible match.
A - 24
Configuration Files
Digit map extension letter “R” indicates that certain matched strings are
replaced. Digit map timer letter “T” indicates a timer expiry. Digit map
protocol letters “S” and “H” indicate the protocol to use when placing a call.
The following examples shows the semantics of the syntax:
• R9RRxxxxxxx—Remove 9 at the beginning of the dialed number

— For example, if a customer dials 914539400, the first 9 is removed
when the call is placed.
• RR604Rxxxxxxx—Prepend 604 to all seven digit numbers

— For example, if a customer dials 4539400, 604 is added to the front of
the number, so a call to 6044539400 is placed.
• R9R604Rxxxxxxx—Replaces 9 with 604
• R999R911R—Convert 999 to 911
• xxR601R600Rxx—When applied on 1160122 gives 1160022
• xR60xR600Rxxxxxxx—Any 60x will be replaced with 600 in the middle of

the dialed number that matches
— For example, if a customer dials 16092345678, a call is placed to
16002345678.
• 911xxx.T— A period (".") which matches an arbitrary number, including

zero, of occurrences of the preceding construct
— For example:
911123 with waiting time to comply with T is a match
and the number can grow indefinitely given that pressing the next
digit takes less than T.
• 0xxxS|33xxH—All four digit numbers starting with a 0 are placed using

the SIP protocol, whereas all four digit numbers starting with 33 are
placed using the H.323 protocol.
Note Only Polycom VVX 1500 phones will match the “H”. It is ignored by all other phones
and the user will need to press the Send soft key to complete dialing. For example,
if the digit map is “33xxH”, the result is as follows:
• If a Polycom VVX 1500 user dials “3302” on an H.323 or dual protocol line, the
call will be placed after the user dials the last digit.
• If a SoundPoint IP 650 user dials “3307”, the user must press the Send soft key
to complete dialing.
The following guidelines should be noted:
• The letters (x, T, R, S, H) are case sensitive.
• You must use only *, #, +, or 0-9 between second and third R
A - 25
• If a digit map does not comply, it is not included in the digit plan as a valid
one. That is, no matching is done against it.
• There is no limitation on the number of R triplet sets in a digit map.

However, a digit map that contains less than full number of triplet sets (for
example, a total of 2Rs or 5Rs) is considered an invalid digit map.
• Using T in the left part of RRR syntax is not recommended. For example,
R0TR322R should be avoided.
Attribute Permitted Values Default Interpretation
dialplan.digitmap string compatible with the [2-9]11|0T| When this attribute is

digit map feature of +011xxx.T| present, number-only dialing
MGCP described in 2.1.5 during the setup phase of
0[2-9]xxxxxxxxx|
of RFC 3435. String is new calls will be compared
limited to 768 bytes and +1[2-9]xxxxxxxx| against the patterns therein
30 segments; a comma is [2-9]xxxxxxxxx| and if a match is found, the
also allowed; when [2-9]xxxT call will be initiated
reached in the digit map, automatically eliminating the
a comma will turn dial need to press Send.
tone back on;’+’ is allowed Attributes
as a valid digit; extension dialplan.applyToCallLis
letter ‘R’ is used as tDial,
defined above. dialplan.applyToDirecto
ryDial,
dialplan.applyToUserDia
l, and
dialplan.applyToUserSen
d control the use of match
and replace in the dialed
number in the different
scenarios.
dialplan.digitmap.timeOut string of positive integers 3|3|3|3|3|3 Timeout in seconds for each

separated by ‘|’ segment of digit map.
Note: If there are more digit
maps than timeout values,
the default value of 3 will be
used. If there are more
timeout values than digit
maps, the extra timeout
values are ignored.
Routing <routing/>
This attribute allows the user to create a specific routing path for outgoing SIP
calls independent of other “default” configurations.
A - 26
Configuration Files
• Emergency <emergency/>
Server <server/>
dialplan.routing.server.x. dotted-decimal IP address Null IP address or host name and port of

address or host name a SIP server that will be used for
routing calls. Multiple servers can
dialplan.routing.server.x.port 1 to 65535 5060 be listed starting with x=1, 2, ... for
fault tolerance.
Emergency <emergency/>
In the following attributes, x is the index of the emergency entry description
and y is the index of the server associated with emergency entry x. For each
emergency entry (index x), one or more server entries (indexes (x,y)) can be
configured. x and y must both use sequential numbering starting at 1.
dialplan.routing.emergency.x. Single entry representing for x =1, This determines the URLs
value a SIP URL value = “911”, Null that should be watched for.
for all others When one of these defined
URLs is detected as having
been dialed by the user, the
call will automatically be
directed to the defined
emergency server.
dialplan.routing.emergency.x. positive integer for x=1, y =1, Null Index representing the
server.y for all others server defined in Server
<server/> on page A-27 that
will be used for emergency
routing.
Localization <lcl/>
The phone has a multilingual user interface. It supports both North American
and international time and date formats. The call progress tones can also be
customized. For more information, refer to Chord-Sets <chord/> on page
A-36, and Call Progress Patterns on page A-40.
• Multilingual <ml/>
• Date and Time <datetime/>
A - 27
Multilingual <ml/>
The multilingual feature is based on string dictionary files downloaded from
the provisioning server. These files are encoded in standalone XML format.
Several eastern European and Asian languages are included with the
distribution.
Attribute Permitted Values Interpretation
lcl.ml.lang Null If Null, the default internal language (US

OR English) will be used, otherwise, the
language to be used may be specified in the
An exact match for
format of lcl.ml.lang.menu.x.label .
one of the label names
stored in For example, to get the phone to boot up in
lcl.ml.lang.menu.x German: lcl.ml.lang = “Deutsch
.label . (de-de)”.
lcl.ml.lang.menu.x String in the format Multiple lcl.ml.lang.menu.x attributes

lcl.ml.lang.menu.x.label language_region are supported - as many languages as are
desired. However, the lcl.ml.lang.menu.x
attributes must be sequential
(lcl.ml.lang.menu.1,
lcl.ml.lang.menu.2,
lcl.ml.lang.menu.3, ...,
lcl.ml.lang.menu.N) with no gaps and the
strings must exactly match a folder name
under the SoundPointIPLocalization folder
on the provisioning server for the phone to
be able to locate the dictionary file.
For example:
lcl.ml.lang.menu.8=“German_Germany”
lcl.ml.lang.menu.8.label=“Deutsch
(de-de)”
lcl.ml.lang.clock.x.24HourClock 0,1 If attribute present, overrides

lcl.datetime.time.24HourClock.
If 1, display time in 24-hour clock mode
rather than am/pm.
lcl.ml.lang.clock.x.format string which includes If attribute present, overrides

‘D’, ‘d’ and ‘M’ and two lcl.datetime.date.format;
optional commas D = day of week
d = day
M = month
Up to two commas may be included.
For example: D,dM = Thursday, 3 July or
Md,D = July 3, Thursday
The field may contain 0, 1 or 2 commas
which can occur only between characters
and only one at a time. For example: “D,,dM”
is illegal.
A - 28
Configuration Files
lcl.ml.lang.clock.x.longFormat 0 or 1 If attribute present, overrides

lcl.datetime.date.longFormat.
If 1, display the day and month in long format
(Friday/November), otherwise use
abbreviations (Fri/Nov).
lcl.ml.lang.clock.x.dateTop 0 or 1 If attribute present, overrides

lcl.datetime.date.dateTop.
If 1, display date above time, otherwise
display time above date.
lcl.ml.lang.y.list “All” or a A list of the languages supported on

comma-separated list hardware platform ‘y’ where ‘y’ can be
IP_500, IP_600, or IP4000.
The IP_500 platform does not support any
Asian languages. The IP_4000 platform does
not support Slovenian.
lcl.ml.lang.tags.x string in the format The format is:

language_region, • The first two letters are the ISO-639
language; preference language abbreviation.
level
• The next two letters are the ISO-3166
country code.
• The next two letters are the ISO-639
language abbreviation.
• The remainder of the string is the
preference level for the display of the
language, or English if the language is
not available
For example:
lcl.ml.lang.tags.1 =
“zh-cn,zh;q=0.9,en;q=0.8”
For more information, refer to the
Accept-Language header definition in the
HTTP RFC 2616 at
http://www.w3.org/Protocols/rfc2616/rfc2616
-sec14.html#sec14.4
To add new languages to those included with the distribution:

1. Create a new dictionary file based on an existing one.
2. Change the strings making sure to encode the XML file in UTF-8 but also
ensuring the UTF-8 characters chosen are within the Unicode character
ranges indicated in the tables below.
3. Place the file in an appropriately named folder according to the format
language_region parallel to the other dictionary files under the
SoundPointIPLocalization folder on the provisioning server.
A - 29
4. Add a lcl.ml.lang.clock.menu.x attribute to the configuration file.

5. Add lcl.ml.lang.clock.x.24HourClock,
lcl.ml.lang.clock.x.format, lcl.ml.lang.clock.x.longFormat
and lcl.ml.lang.clock.x.dateTop attributes and set them according
to the regional preferences.
6. (Optional) Set lcl.ml.lang to be the new language_region string.
Basic character support includes the following Unicode

character ranges
Name Range
C0 Controls and Basic Latin U+0000 - U+007F
C1 Controls and Latin-1 Supplement U+0080 - U+00FF
Cyrillic (partial) U+0400 - U+045F
Extended character support available on SoundPoint IP 600 and SoundStation IP 4000 and 7000 platforms
includes the following Unicode character ranges
Name Range
CJK Symbols and Punctuation U+3000 - U+303F
Hiragana U+3040 - U+309F
Katakana U+30A0 - U+30FF
Bopomofo U+3100 - U+312F
Hangul Compatibility Jamo U+3130 - U+318F
Bopomofo Extended U+31A0 - U+31BF
Enclosed CJK Letters and Months U+3200 - U+327F
CJK Compatibility U+3300 - U+33FF
CJK Unified Ideographs U+4E00 - U+9FFF
Hangul Syllables U+AC00 - U+D7A3
CJK Compatibility Ideographs U+F900 - U+FAFF
CJK Half-width forms U+FF00 - U+FFFF
Note Within a Unicode range, some characters may not be supported due to their
infrequent usage
A - 30
Configuration Files
Date and Time <datetime/>

Permitted
Attribute Values Interpretation
lcl.datetime.time.24HourClock 0,1 If set to 1, display time in 24-hour clock mode rather

than a.m./p.m.
lcl.datetime.date.format string which Controls format of date string.

includes ‘D’, ‘d’ D = day of week
and ‘M’ and two d = day
optional commas M = month
Up to two commas may be included.
For example: D,dM = Thursday, 3 July or Md,D = July
3, Thursday
The field may contain 0, 1 or 2 commas which can
occur only between characters and only one at a time.
For example: “D,,dM” is illegal.
lcl.datetime.date.longFormat 0,1 If set to 1, display the day and month in long format
(Friday/November), otherwise, use abbreviations
(Fri/Nov).
lcl.datetime.date.dateTop 0 or 1 If set to 1, display date above time else display time

above date.
User Preferences <up/>

Permitted
up.headsetMode 0 or 1 0 If set to 1, the headset will be selected as the

preferred transducer after its first use until the
headset key is pressed again; otherwise,
hands-free will be selected preferentially over
the headset.
up.useDirectoryNames 0 or 1 0 If set to 1, the name fields of the local contact

directory entries which match incoming calls
will be used for caller identification display and
in the call lists instead of the name provided
through network signaling.
Note: There is no matching of outgoing calls.
There is no matching to corporate directory
entries.
up.oneTouchVoiceMail 0 or 1 0 If set to 1, the voice mail summary display is

bypassed and voice mail is dialed directly (if
configured).
A - 31
Permitted
up.welcomeSoundEnabled 0 or 1 1 If set to 1, play welcome sound effect after a

reboot.
up.welcomeSoundOnWarmBootE 0 or 1 0 If set to 1, play welcome sound effect on warm

nabled and cold boots.
If set to 0, only a cold reboot will trigger the
welcome sound effect.
up.localClockEnabled 0 or 1 1 If set to 1, display the date and time on the idle

display.
up.backlight.onIntensity 0 (off), Null This parameter controls the intensity of the

1 (low), LCD backlight when it turns on during normal
2 (medium), 3 use of the phone.
(high) The default value is medium.
up.backlight.idleIntensity 0 (off), Null This parameter controls the intensity of the

1 (low), LCD backlight when the phone is idle.
2 (medium), 3 The default value is low.
(high)
Note: If idleIntensity is set higher than
onIntensity, it will be replaced with the
onIntensity value.
up.toneControl.bass -4 to 4, Null 0 Bass equalization control.

Each step is an increment of 1 dB at 225 kHz
and 2 dB < 225 Hz.
up.toneControl.treble -4 to 4, Null 0 Treble equalization control.

Each step is an increment of 1 dB at 3.7 kHz
and 2 dB > 10 kHz.
up.audioSetup.auxInput 0 - Other Null Auxiliary audio input.

Input, If set to Null, default value is 2.
1 - Polycom
Wireless Mic,
2 - off
up.audioSetup.auxOutput 0 - Other Null Auxiliary audio output.

Input, If set to Null, default value is 2.
1 - Polycom
Wireless Mic,
2 - off
up.idleTimeout positive Null Timeout for the idle display or default call
integer, handling display.
seconds If set to 0, there is no timeout.
If set to Null, the default timeout of 40 seconds
is used.
If set to value greater than 0, the timeout is for
that number of seconds (maximum 65535).
A - 32
Configuration Files
Permitted
up.mwiVisible 0 or 1 0 If set is 0 or Null, the incoming MWI

notifications for lines where the MWI callback
mode is disabled (msg.mwi.x.callBackMode
is set to 0) are ignored, and do not appear in
the message retrieval menus.
If set to 1, the MWI for lines whose MWI is
disabled is displayed (pre-SIP 2.1 behavior),
even though MWI notifications have been
received for those lines.
up.handsfreeMode 0 or 1 1 If set to 1 or Null, hands-free speakerphone is

enabled.
If set to 0, hands-free speakerphone is
disabled.
up.numberFirst CID 0 or 1 0 If set to 0 or Null, caller ID display will show

caller’s name first.
If set to 1, caller ID display will show caller’s
number first.
up.idleBrowser.enabled 0 or 1 Null A flag to determine whether or not the

background takes priority over the idle
browser. Used in conjunction with
up.prioritizeBackground.enable .
up.prioritizeBackgroundMenuItem 0 or 1 1 If set to 1, the “Prioritize Background” menu is

.enable available to the user. The user can then decide
whether or not the background takes priority
over the idle browser. Used in conjunction with
up.idleBrowser.enabled .
up.screenCapture.enabled 0 or 1 0 A flag to determine whether or not the user

can get a screen capture of the current screen
shown on a phone. The flag is cleared when
the phone reboots.
If set to 1, the “Screen Capture” menu is
available to the user.
Refer to Capturing Phone’s Current Screen on
page C-30.
up.manualProtocolRouting 0 or 1 1 If set to 1, the user is presented with protocol

routing choices when a call could be placed
with more than one protocol from its current
context. The user must choose between SIP
and H.323 to place a call.
Note: This parameter is supported on the
Polycom VVX 1500 only.
A - 33
Permitted
up.manualProtocolRouting. 0 or 1 1 If set to 1 and up.manualProtocolRouting is

softKeys set to 1, soft keys are used to provide the user
with a routing choice.
If set to 0, a routing confirmation dialog is
presented with a choice for each possible
routing.
up.callTypePromptPref 0 or 1 1 A flag to determine which interface is used to

place calls.
If set to 1 or Null, the voice interface is used.
If set to 0, the video interface is used to place
calls during hot-dialing or if the use presses
the off-hook key on the SoundStation IP 7000
phone.
SoundStation IP 7000 only.
up.enableCallTypePrompt 0 or 1 1 Enable the call type prompt.

If set to 1 or Null, the call type prompt is
enabled.
If set to 0, the call type prompt is disabled.
SoundStation IP 7000 only.
Tones <tones/>
This attribute describes configuration items for the tone resources available in
the phone.
• Dual Tone Multi-Frequency <DTMF/>
• Chord-Sets <chord/>
A - 34
Configuration Files
Dual Tone Multi-Frequency <DTMF/>

Permitted
tone.dtmf.level -33 to -3 -15 Level of the high frequency component of

the DTMF digit measured in dBm0; the
low frequency tone will be two dB lower.
tone.dtmf.onTime positive 50 When a sequence of DTMF tones is

integer played out automatically, this is the length
of time in milliseconds the tones will be
generated for; this is also the minimum
time the tone will be played for when
dialing manually (even if key press is
shorter).
tone.dtmf.offTime positive 50 When a sequence of DTMF tones is

integer played out automatically, this is the length
of time in milliseconds the phone will
pause between digits; this is also the
minimum inter-digit time when dialing
manually.
tone.dtmf.chassis.masking 0 or 1 0 If set to 1, DTMF tones will be substituted

with a non-DTMF pacifier tone when
dialing in hands-free mode. This prevents
DTMF digits being broadcast to other
surrounding telephony devices or being
inadvertently transmitted in-band due to
local acoustic echo.
Note: tone.dtmf.chassis.masking should
only be enabled when tone.dtmf.viaRtp is
disabled.
tone.dtmf.stim.pac.offHookOnly 0 or 1 0 Not currently used.
A - 35
Permitted
tone.dtmf.viaRtp 0 or 1 1 If set to 1, encode DTMF in the active

RTP stream, otherwise, DTMF may be
encoded within the signaling protocol only
when the protocol offers the option.
Note: tone.dtmf.chassis.masking should
be enabled when tone.dtmf.viaRtp is
disabled.
tone.dtmf.rfc2833Control 0 or 1 1 If set to 1, the phone will indicate a

preference for encoding DTMF through
RFC 2833 format in its Session
Description Protocol (SDP) offers by
showing support for the phone-event
payload type; this does not affect SDP
answers, these will always honor the
DTMF format present in the offer since
the phone has native support for RFC
2833.
tone.dtmf.rfc2833Payload 96-127 127 The phone-event payload encoding in the

dynamic range to be used in SDP offers.
Chord-Sets <chord/>
Chord-sets are the building blocks of sound effects that use synthesized rather
than sampled audio (most call progress and ringer sound effects). A chord-set
is a multi-frequency note with an optional on/off cadence. A chord-set can
contain up to four frequency components generated simultaneously, each
with its own level.
There are three blocks of chord sets:
• callProg (used for call progress sound effect patterns)
• ringer
• misc (miscellaneous)
All three blocks use the same chord set specification format.
A - 36
Configuration Files
In the following table, x is the chord-set number and cat is one of callProg,
ringer, or misc.
Permitted
tone.chord.cat.x.freq.y 0-1600 Frequency for this component in Hertz; up to four

chord-set components can be specified (y=1, 2, 3, 4).
tone.chord.cat.x.level.y -57 to 3 Level of this component in dBm0.
tone.chord.cat.x.onDur positive On duration in milliseconds, 0=infinite.

integer
tone.chord.cat.x.offDur positive Off duration in milliseconds, 0=infinite.

integer
tone.chord.cat.x.repeat positive Specifies how many times the ON/OFF cadence is

integer repeated, 0=infinite.
Sampled Audio for Sound Effects <saf/>

The following sampled audio WAVE file (.wav) formats are supported:
• mono 8 kHz G.711 μ-Law
• G.711 A-Law
• L16/16000 (16-bit, 16 kHz sampling rate, mono)
Note L16/32000 and L16/48000 are supported on SoundStation IP 6000 and 7000
phones.
The phone uses built-in wave files for some sound effects. The built-in wave
files can be replaced with files downloaded from the provisioning server or
from the Internet, however, these are stored in volatile memory so the files will
need to remain accessible should the phone need to be rebooted. Files will be
truncated to a maximum size of 300 kilobytes.
A - 37
In the following table, x is the sampled audio file number.
saf.x Null OR valid path name If Null, the phone will use a built-in file.
OR an RFC If set to a path name, the phone will attempt to download this file
1738-compliant URL to a at boot time from the provisioning server.
HTTP, FTP, or TFTP wave
If set to a URL, the phone will attempt to download this file at boot
file resource.
time from the Internet.
Note: Refer to the above
Note: A TFTP URL is expected to be in the format:
wave file format
tftp://<host>/[pathname]<filename>, for example:
restrictions.
tftp://somehost.example.com/sounds/example.wav .
The following table defines the default usage of the sampled audio files with
the phone:
Sampled Audio File Default use within phone (pattern reference)

1 Ringer 12 (se.pat.misc.7)
2 Ringer 13 (se.pat.ringer.13)
12-24 Not used.
Note In SIP 3.1, the SoundPoint IP welcome sound was removed from saf.1 . If you
want the welcome sound to be played when a phone reboots or restarts, set saf.1
to SoundPointIPWelcome.wav .
Sound Effects <se/>

The phone uses both synthesized (based on the chord-sets, refer to Chord-Sets
<chord/> on page A-36) and sampled audio sound effects. Sound effects are
defined by patterns: rudimentary sequences of chord-sets, silence periods, and
wave files.
A - 38
Configuration Files
Permitted
se.stutterOnVoiceMail 0 or 1 1 If set to 1, stuttered dial tone is used in place of

normal dial tone to indicate that one or more
messages (voice mail) are waiting at the message
center.
se.appLocalEnabled 0 or 1 1 If set to 1, local user interface sound effects such

as confirmation/error tones, will be enabled.
• Patterns <pat/>
• Ring type <rt/>
Patterns <pat/>
Patterns use a simple script language that allows different chord sets or wave
files to be strung together with periods of silence. The script language uses the
following instructions:
Instruction Meaning Example
sampled (n) Play sampled audio file se.pat.callProg.x.inst.y.type =”sampled” (sampled audio
n file instruction type)
se.pat.callProg.x.inst.y.value =”3” (specifies sampled
audio file 3)
chord (n, d) Play chord set n (d is se.pat.callProg.x.inst.y.type = “chord” (chord set

optional and allows the instruction type)
chord set ON duration to se.pat.callProg.x.inst.y.value = “3” (specifies call
be overridden to d progress chord set 3)
milliseconds)
se.pat.callProg.x.inst.y.param = “2000” (override ON
duration of chord set to 2000 milliseconds)
silence (d) Play silence for d se.pat.callProg.x.inst.y.type = “silence” (silence

milliseconds (Rx audio instruction type)
is not muted) se.pat.callProg.x.inst.y.value = “300” (specifies silence is
to last 300 milliseconds)
branch (n) Advance n instructions se.pat.callProg.x.inst.y.type = “branch” (branch

and execute that instruction type)
instruction (n must be se.pat.callProg.x.inst.y.value = “-5” (step back 5
negative and must not instructions and execute that instruction)
branch beyond the first
instruction)
A - 39
Note Currently, patterns that use the sampled instruction are limited to the following
format: sampled followed by optional silence and optional branch back to the
beginning.
In the following table, x is the pattern number, y is the instruction number.

Both x and y need to be sequential. There are three categories of sound effect
patterns: callProg (Call Progress Patterns), ringer (Ringer Patterns) and misc
(Miscellaneous Patterns).
Permitted
se.pat.callProg.x.name UTF-8 Used for identification purposes in the user interface (currently
encoded used for ringer patterns only); for patterns that use a sampled
string audio file which has been overridden by a downloaded
replacement, the se.pat.ringer.x.name parameter will be
overridden in the user interface by the file names of the wave file.
se.pat.callProg.x.inst.y.type sampled OR As above.

chord OR
silence OR
branch
se.pat.callProg.x.inst.y. integer Instruction type: Interpretation:

value sampled sampled audio file number
chord chord set number
silence silence duration in ms
branch number of instructions to advance
se.pat.callProg.x.inst.y. positive If instruction type is chord, this optional parameter specifies the on
param integer duration to be used, overriding the on duration specified in the
chord-set definition.
Call Progress Patterns

The following table maps call progress patterns to their usage within the
phone.
Call progress
pattern number Use within phone
1 dial tone
2 busy tone
3 ring back tone
4 reorder tone
5 stuttered dial tone
6 call waiting tone
A - 40
Configuration Files
Call progress
7 alternate call waiting tone (distinctive)
8 confirmation tone
9 howler tone (off-hook warning)
10 record warning
11 message waiting tone
12 alerting
13 intercom announcement tone
14 barge-in tone
15 secondary dial tone
Ringer Patterns
The following table maps ringer pattern numbers to their default descriptions.
Ringer pattern number Default description
1 Silent Ring
2 Low Trill
3 Low Double Trill
4 Medium Trill
5 Medium Double Trill
6 High Trill
7 High Double Trill
8 Highest Trill
9 Highest Double Trill
10 Beeble
11 Triplet
12 Ringback-style
13 Sampled audio file 1
A - 41
Ringer pattern number Default description
Note Silent Ring will only provide a visual indication of an incoming call, but no audio
indication.
Sampled audio files 1-21 all use the same built-in file unless that file has been
replaced with a downloaded file. For more information, refer to Sampled Audio for
Sound Effects <saf/> on page A-37.
Miscellaneous Patterns
The following table maps miscellaneous patterns to their usage within the
phone.
Miscellaneous
1 new message waiting indication
2 new instant message
3 Not used
4 local hold notification
5 positive confirmation
6 negative confirmation
7 welcome (boot up)
A - 42
Configuration Files
Ring type <rt/>

Ring type is used to define a simple class of ring to be applied based on some
credentials that are usually carried within the network protocol. The ring class
includes attributes such as call-waiting and ringer index, if appropriate. The
ring class can use one of four types of ring that are defined as follows:
ring Play a specified ring pattern or call waiting indication.

visual Provide only a visual indication (no audio indication) of incoming call (no
ringer needs to be specified).
answer Provide auto-answer on incoming call.
ring-answer Provide auto answer on incoming call after a ring period.
Note The auto-answer on incoming call is currently only applied if there is no other call in
progress on the phone at the time.
In the following table, x is the ring class number. The x index needs to be
sequential.
se.rt.enabled 0,1 Set to 1 to enable the ring type feature within the
phone, 0 otherwise.
se.rt.modification.enabled 0,1 Set to 1 to allow user modification through local

user interface of the pre-defined ring type enabled
for modification.
se.rt.x.name UTF-8 encoded string Used for identification purposes in the user
interface.
se.rt.x.type ring OR visual OR answer As defined in table above.

OR ring-answer
se.rt.x.ringer integer - only relevant if the The ringer index to be used for this class of ring.
type is set to ‘ring’ or The ringer index should match one of Ringer
‘ring-answer’ Patterns on page A-41.
se.rt.x.callWait integer - only relevant if the The call waiting index to be used for this class of
type is set to ‘ring’ or ring. The call waiting index should match one
‘ring-answer’ defined in Call Progress Patterns on page A-40.
se.rt.x.timeout positive integer - only The duration of the ring in milliseconds before the
relevant if the type is set to call is auto answered. If this field is omitted or is left
‘ring-answer’. Default blank, a value of 2000 is used.
value is 2000.
se.rt.x.mod 0,1 Set to 1 if the user interface should allow for

modification by the user of the ringer index used for
this ring class.
A - 43
Note Modification of se.rt.modification.enabled and se.rt.x.name parameters

through the user interface will be implemented in a future release.
Voice Settings <voice/>

Permitted
voice.txPacketFilter 0 or 1 Null Flag to determine whether or not narrowband Tx

high-pass filtering should be enabled.
If set to 1, narrowband Tx high-pass filter is
enabled.
If set 0 or Null, no Tx filtering is performed.
• Voice Coding Algorithms <codecs/>
• Volume Persistence <volume/>
• Gains <gain/>
• Acoustic Echo Cancellation <aec/>
• Acoustic Echo Suppression <aes/>
• Background Noise Suppression <ns/>
• Automatic Gain Control <agc/>
• Receive Equalization <rxEq/>
• Transmit Equalization <txEq/>
• Voice Activity Detection <vad/>
• Quality Monitoring <quality monitoring/>
Voice Coding Algorithms <codecs/>

These codecs include:
• Codec Preferences <codecPref/>
• Codec Profiles <audioProfile/>
A - 44
Configuration Files
Codec Preferences <codecPref/>
Permitted
voice.codecPref.G711Mu Null, 1-3 1 Specifies the codec preferences for

SoundPoint IP 32x/33x, and 430 phones.
voice.codecPref.G711A 2
1 = highest
voice.codecPref.G729AB 3 3 = lowest
Null = do not use
voice.codecPref.iLBC.13_33kbps Null Give each codec a unique priority, this will
voice.codecPref.iLBC.15_2kbps Null dictate the order used in SDP negotiations.
Note: iLBC is not supported on the
SoundPoint IP 430.
voice.codecPref.IP_650.G711Mu Null, 1-4 2 Specifies the codec preferences for the

SoundPoint IP 450, 550, 560, 650, and
voice.codecPref.IP_650.G711A 3 670 phones. Interpretation as above.
voice.codecPref.IP_650.G729AB 4
voice.codecPref.IP_650.G722 1
voice.codecPref.iLBC.IP_650. Null
13_33kbps
voice.codecPref.iLBC.IP_650.15_2kbps Null
A - 45
Permitted

SoundStation IP 6000 phone.
voice.codecPref.IP_6000.G711A 6 Interpretation as above.
voice.codecPref.IP_6000.G7221.16kbps Null
voice.codecPref.IP_6000.G7221.32kbps 4
voice.codecPref.IP_6000.G7221C. Null
24kbps
32kbps
voice.codecPref.IP_6000.G7221C. 1
48kbps
voice.codecPref.IP_6000.Siren14. Null
24kbps
32kbps
voice.codecPref.IP_6000.Siren14. 2
48kbps
13_33kbps
15_2kbps
A - 46
Configuration Files
Permitted

SoundStation IP 7000 phone.
voice.codecPref.IP_7000.G711A 7 Interpretation as above.
voice.codecPref.IP_7000.G7221.32kbps 5
24kbps
32kbps
voice.codecPref.IP_7000.G7221C. 2
48kbps
voice.codecPref.IP_7000.Lin16.16ksps Null
32kbps
48kbps
64kbps
24kbps
32kbps
48kbps
13_33kbps
15_2kbps
A - 47
Permitted
voice.codecPref.VVX_1500.G711Mu Null, 1-16 4 Specifies the audio codec preferences for

the Polycom VVX 1500 phone.
voice.codecPref.VVX_1500.G711A 5 Interpretation as above.
voice.codecPref.VVX_1500. Null
G719.32kbps
G719.48kbps
G719.64kbps
voice.codecPref.VVX_1500.G722 3
G7221.16kbps
G7221.24kbps
voice.codecPref.VVX_1500. 2
G7221.32kbps
G7221C.24kbps
G7221C.32kbps
voice.codecPref.VVX_1500. 1
G7221C.48kbps
voice.codecPref.VVX_1500.G729AB 6
Lin16.16ksps
Lin16.32ksps
Lin16.44_1ksps
Lin16.48ksps
voice.codecPref.VVX_1500.Lin16.8ksps Null
Siren14.24kbps
Siren14.32kbps
A - 48
Configuration Files
Permitted
Siren14.48kbps
voice.codecPref.iLBC.VVX_1500. Null
13_33kbps
voice.codecPref.iLBC.VVX_1500. Null
15_2kbps
Note Codecs with a default of Null are available for test purposes only and are not
expected to be used in your deployment.
Codec Profiles <audioProfile/>

The following profile attributes can be adjusted for each of the five supported
codecs. In the table, x=G711Mu, G711A, G719, G722, G7221, G7221C, G729AB,
Lin16, Siren14, Siren22, and iLBC.
Permitted
voice.audioProfile.x.payloadSize 10, 20, 30, ...80 Preferred Tx payload size in milliseconds to be

provided in SDP offers and used in the
absence of ptime negotiations. This is also the
range of supported Rx payload sizes.
The payload size for G719, G7221, G7221C,
Siren14, Siren22, and iLBC are further
subdivided.
voice.audioProfile.x.jitterBufferMin 20, 40, 50, 60, The smallest jitter buffer depth (in milliseconds)
... (multiple of that must be achieved before play out begins
10) for the first time. Once this depth has been
achieved initially, the depth may fall below this
point and play out will still continue. This
parameter should be set to the smallest
possible value which is at least two packet
payloads, and larger than the expected short
term average jitter. The IP4000 values are the
same as the IP30x values.
voice.audioProfile.x.jitterBufferShrink 10, 20, 30, ... The absolute minimum duration time (in
(multiple of 10) milliseconds) of RTP packet Rx with no packet
loss between jitter buffer size shrinks. Use
smaller values (1000 ms) to minimize the delay
on known good networks. Use larger values to
minimize packet loss on networks with large
jitter (3000 ms).
A - 49
Permitted
voice.audioProfile.x.jitterBufferMax > The largest jitter buffer depth to be supported

jitterBufferMin, (in milliseconds). Jitter above this size will
multiple of 10, always cause lost packets. This parameter
<=300 for IP should be set to the smallest possible value
32x, 33x, 430, that will support the expected network jitter.
550, 600, and
650
voice.audioProfile.x.payloadType 96 - 127 The codec payload encoding in the dynamic

(default) range to be used in SDP offers.
Volume Persistence <volume/>

The user’s selection of the receive volume during a call can be remembered
between calls. This can be configured per termination (handset, headset and
hands-free/chassis). In some countries regulations exist which dictate that
receive volume should be reset to nominal at the start of each call on handset
and headset.
Permitted
voice.volume.persist.handset 0 or 1 0 If set to 1, the receive volume will be

remembered between calls.
voice.volume.persist.headset 0 or 1 0
If set to 0, the receive volume will be reset
voice.volume.persist.handsfree 0 or 1 1 to nominal at the start of each call.
Gains <gain/>
The default gain settings have been carefully adjusted to comply with the
TIA-810-A digital telephony standard.
Polycom recommends that you do not change these values.
Attribute Default
voice.gain.rx.analog.handset 0
voice.gain.rx.analog.handset.VVX_1500 -2
voice.gain.rx.analog.headset 0
voice.gain.rx.analog.headset.VVX_1500 -2
voice.gain.rx.analog.chassis 0
A - 50
Configuration Files
Attribute Default
voice.gain.rx.analog.chassis.IP_330 0
voice.gain.rx.analog.chassis.VVX_1500 -3
voice.gain.rx.analog.ringer 0
voice.gain.rx.analog.ringer.IP_330 0
voice.gain.rx.analog.ringer.VVX_1500 0
voice.gain.rx.digital.handset -15
voice.gain.rx.digital.headset -21
voice.gain.rx.digital.chassis 0
voice.gain.rx.digital.chassis.IP_450 5
voice.gain.rx.digital.chassis.VVX_1500 0
voice.gain.rx.digital.ringer -21
voice.gain.rx.digital.ringer.IP_330 -12
voice.gain.rx.digital.ringer.VVX_1500 -21
A - 51
Attribute Default
voice.gain.rx.analog.handset.sidetone -20
voice.gain.rx.analog.handset.sidetone.VVX_1500 -15
voice.gain.rx.analog.headset.sidetone -24
voice.gain.rx.analog.headset.sidetone.VVX_1500 -31
voice.gain.tx.analog.handset 6
voice.gain.tx.analog.handset.VVX_1500 -48
voice.gain.tx.analog.headset 3
voice.gain.tx.analog.headset.VVX_1500 -47
voice.gain.tx.analog.chassis 3
voice.gain.tx.analog.chassis.IP_330 36
voice.gain.tx.analog.chassis.VVX_1500 -25
voice.gain.tx.digital.handset 0
voice.gain.tx.digital.handset.IP_330 10
voice.gain.tx.digital.handset.VVX_1500 12
voice.gain.tx.digital.headset 0
voice.gain.tx.digital.headset.IP_330 10
voice.gain.tx.digital.headset.VVX_1500 12
voice.gain.tx.digital.chassis 3
voice.gain.tx.digital.chassis.IP_330 12
A - 52
Configuration Files
Attribute Default
voice.gain.tx.digital.chassis.VVX_1500 3
voice.gain.tx.analog.preamp.handset 23
voice.gain.tx.analog.preamp.headset 23
voice.gain.tx.analog.preamp.chassis 32
voice.gain.tx.analog.preamp.chassis.IP_601 32
voice.handset.rxag.adjust.IP_330 1
voice.handset.txag.adjust.IP_330 18
voice.handset.sidetone.adjust.IP_330 3
voice.headset.rxag.adjust.IP_330 4
voice.headset.txag.adjust.IP_330 21
A - 53
Attribute Default
voice.headset.sidetone.adjust.IP_330 -3
Acoustic Echo Cancellation <aec/>

These settings control the performance of the speakerphone acoustic echo
canceller.
Attribute Default
voice.aec.hs.enable 1
voice.aec.hs.lowFreqCutOff 100
voice.aec.hs.highFreqCutOff 7000
voice.aec.hs.erlTab_0_300 -24
voice.aec.hd.enable 0
voice.aec.hd.lowFreqCutOff 100
voice.aec.hd.highFreqCutOff 7000
voice.aec.hd.erlTab_0_300 -24
voice.aec.hf.enable 1
voice.aec.hf.lowFreqCutOff 100
A - 54
Configuration Files
Attribute Default
voice.aec.hf.highFreqCutOff 7000
voice.aec.hf.erlTab_0_300 -6
Acoustic Echo Suppression <aes/>

Acoustic Echo Suppression (AES) provides non-linear processing of the
microphone signal to remove any residual echo remaining after linear AEC
processing. Because AES depends on AEC, AES should only be enabled when
AEC is also enabled. Normally, AES should be used whenever AEC is used for
handsfree or handset and both are enabled by default for those terminations
These settings control the performance of the speakerphone acoustic echo
suppressor.
Attribute Default
voice.aes.hs.enable 1
voice.aes.hs.duplexBalance 7
voice.aes.hd.enable 0
voice.aes.hd.duplexBalance 0
voice.aes.hf.enable 1
voice.aes.hf.duplexBalance.0 7
A - 55
Attribute Default
voice.aes.hf.duplexBalance.IP_4000.0 10
Background Noise Suppression <ns/>

These settings control the performance of the transmit background noise
suppression feature.
Attribute Default
voice.ns.hs.enable 1
voice.ns.hs.signalAttn -6
voice.ns.hs.silenceAttn -9
voice.ns.hd.enable 0
voice.ns.hd.signalAttn 0
voice.ns.hd.silenceAttn 0
voice.ns.hf.enable 1
voice.ns.hf.signalAttn -6
voice.ns.hf.silenceAttn -9
A - 56
Configuration Files
Attribute Default
voice.ns.hf.IP_4000.enable 1
voice.ns.hf.IP_4000.signalAttn -6
voice.ns.hf.IP_4000.silenceAttn -9
Automatic Gain Control <agc/>

These settings control the performance of the transmit automatic gain control
feature.
Note Automatic Gain Control will be implemented in a future release.
Attribute Default
voice.agc.hs.enable 0
voice.agc.hd.enable 0
voice.agc.hf.enable 0
Receive Equalization <rxEq/>

These settings control the performance of the receive equalization feature.
Attribute Default
voice.rxEq.hs.IP_330.preFilter.enable 1
voice.rxEq.hs.VVX_1500.preFilter.enable 1
A - 57
Attribute Default
voice.rxEq.hs.IP_330.postFilter.enable 0
voice.rxEq.hs.VVX_1500.postFilter.enable 0
voice.rxEq.hd.IP_330.preFilter.enable 0
voice.rxEq.hd.VVX_1500.preFilter.enable 0
voice.rxEq.hd.IP_330.postFilter.enable 0
voice.rxEq.hd.VVX_1500.postFilter.enable 0
voice.rxEq.hf.IP_330.preFilter.enable 1
voice.rxEq.hf.VVX_1500.preFilter.enable 1
voice.rxEq.hf.IP_330.postFilter.enable 0
voice.rxEq.hf.VVX_1500.postFilter.enable 0
A - 58
Configuration Files
Transmit Equalization <txEq/>

These settings control the performance of the hands-free transmit equalization
feature.
Attribute Default
voice.txEq.hs.IP_330.preFilter.enable 0
voice.txEq.hs.VVX_1500.preFilter.enable 0
voice.txEq.hs.IP_330.postFilter.enable 1
voice.txEq.hs.VVX_1500.postFilter.enable 1
voice.txEq.hd.IP_330.preFilter.enable 0
voice.txEq.hd.VVX_1500.preFilter.enable 0
voice.txEq.hd.IP_330.postFilter.enable 0
voice.txEq.hd.VVX_1500.postFilter.enable 0
voice.txEq.hf.IP_330.preFilter.enable 0
A - 59
Attribute Default
voice.txEq.hf.VVX_1500.preFilter.enable 0
voice.txEq.hf.IP_330.postFilter.enable 1
voice.txEq.hf.VVX_1500.postFilter.enable 1
Voice Activity Detection <vad/>

These settings control the performance of the voice activity detection (silence
suppression) feature.
Permitted
voice.vadEnable 0 or 1 0 If set to 1, enable VAD.
voice.vadThresh integer from 0 15 The threshold for determining what is active voice and
to 30 what is background noise in dB. This does not apply to
G.729AB codec operation which has its own built-in VAD
function.
voice.vad. 0 or 1 Null If set to 1 or Null and voice.vadEnable is set to 1,

signalAnnexB Annex B is used. A new line can be added to SDP
depending on the setting of this parameter and the
voice.vadEnable parameter.
• If voice.vadEnable is set to 1, add attribute line
a=fmtp:18 annexb="yes" below a=rtpmap… attribute
line (where '18' could be replaced by another
payload).
• If voice.vadEnable is set to 0, add attribute line
a=fmtp:18 annexb="no" below a=rtpmap… attribute
line (where '18' could be replaced by another
payload).
If set to 0, there is no change to SDP.
A - 60
Configuration Files
Quality Monitoring <quality monitoring/>

• Central Report Collector <collector/>
• Alert Reports <alert/>
• RTCP-XR <rtcpxr/>
Central Report Collector <collector/>
Permitted
voice.qualityMonitoring.collector.enable. 0, 1 0 Enables generation of periodic

periodic quality reports throughout a call.
voice.qualityMonitoring.collector.enable. 0, 1 0 Enables generation of a quality

session report at the end of each call.
voice.qualityMonitoring.collector.enable. 0, 1, 2 0 Controls the generation of periodic

triggeredPeriodic quality reports triggered by alert
states.
If set to 0, alert states do not cause
periodic reports to be generated.
If set to 1, periodic reports will be
generated when an alert state is
critical.
If set to 2, periodic reports will be
generated when an alert state is
either warning or critical.
Note: This parameter is ignored
when
qualityMonitoring.collector.e
nable.periodic is set 1, since
periodic reports are sent throughout
the duration of a call.
voice.qualityMonitoring.collector.period 5 to 20 20 The time interval between

successive periodic quality reports.
A - 61
Alert Reports <alert/>

Permitted
voice.qualityMonitoring.collector. Null, 15 to 40 Null Threshold value of listening MOS

alert.moslq.threshold.warning score (MOS-LQ) that causes phone
to send a warning alert quality
report. Configure the desired MOS
value multiplied by 10. If set to Null,
warning alerts are not generated
due to MOS-LQ.
For example, a configured value of
35 corresponds to the MOS score
3.5.
voice.qualityMonitoring.collector. Null, 15 to 40 Null Threshold value of listening MOS

alert.moslq.threshold.critical score (MOS-LQ) that causes phone
to send a critical alert quality report.
Configure the desired MOS value
multiplied by 10. If set to Null,
critical alerts are not generated due
to MOS-LQ.
For example, a configured value of
28 corresponds to the MOS score
2.8.
voice.qualityMonitoring.collector.alert. Null, 10 to Null Threshold value of one way delay

delay.threshold.warning 2000 (in ms) that causes phone to send a
critical alert quality report. If set to
Null, warning alerts are not
generated due to one way delay.
One-way delay includes both
network delay and end system
delay.
voice.qualityMonitoring.collector.alert. Null, 10 to Null Threshold value of one way delay

delay.threshold.critical 2000 (in ms) that causes phone to send a
critical alert quality report. If set to
Null, critical alerts are not generated
due to one way delay. One-way
delay includes both network delay
and end system delay.
A - 62
Configuration Files
Server <server/>
Permitted
voice.qualityMonitoring.collector.server.x. Dotted-decima Null IP address or host name and port of

address l IP address or a SIP server (report collector) that
host name accepts voice quality reports
contained in SIP PUBLISH
messages. Set x to 1as only one
report collector is supported at this
time.
voice.qualityMonitoring.collector.server.x. 0, Null, 1 to 5060 If port is 0 or Null, port 5060 will be

port 65535 used. Set x to 1as only one report
collector is supported at this time.
RTCP-XR <rtcpxr/>
Permitted
voice.qualityMonitoring.rtcpxr.enable 0, 1 0 Enables generation of RTCP-XR

packets.
A - 63
Video Settings <video/>
Note This attribute is only supported for use on the Polycom VVX 1500.
These configuration attributes are defined as follows:
Permitted
video.enable 0=Disable, 1 Flag to determine whether or not video calls are

1=Enable established. This applies to all calls, between two
Polycom VVX 1500s and between
Polycom VVX 1500 and any other video device.
If set to 1 (enabled) or Null, video is sent in
outgoing calls and received in incoming calls.
If set to 0, video is not sent in outgoing calls and
not received in incoming calls. All calls are audio
only.
video.autoStartVideoTx 0 or 1 1 Flag to determine whether or not video

transmission occurs when a call starts.
If set to 0, video transmission does not start.
If set to 1or Null, video transmission from the near
end starts when a call starts.
video.screenMode “normal”, “full”, normal Applies to the video window shown in the normal
“crop” mode.
If set to “normal” or Null, all pixels are displayed,
black bars appear on the top, bottom, or sides of
the window, if necessary, to maintain the correct
aspect ratio.
If set to “full”, all pixels are displayed and the
image is stretched linearly and independently to fill
the video frame.
If set to “crop”, the black bars do not appear, the
image size is re-sized to maintain the correct
aspect ratio, and any parts of the image that do not
fit in the display are cropped.
video.screenModeFS “normal”, “full”, normal Applies to the video window in Full Screen mode.
“crop” The image is re-sized to maintain the correct
aspect ratio and any parts of the image that do not
fit in the display are cropped.
video.quality “motion”, Null Determine the quality of video shown in a call or

“sharpness” conference.
Use “motion” for people or other video with motion.
Use “sharpness” or Null for video with little or no
movement. Moderate to heavy motion can cause
some frames to be dropped.
A - 64
Configuration Files
Permitted
video.callRate 128 - 1024 448 The maximum call rate in kbps to use when initially
kbps negotiating the bandwidth for a video call.
This value cannot exceed video.maxCallRate.
video.maxCallRate 128 - 1024 512 Limits the maximum network bandwidth used in a
kbps call. It is used in the SDP bandwidth signaling.
If honored by the far end, both Rx and Tx network
bandwidth used in a call will not exceed this value
(in kbps).
If set to Null, the value 1024 is used.
video.autoFullScreen 0 or 1 Null Flag to determine whether or not video calls use

the full screen layout.
If set to 1, video calls will use the full screen layout
by default. When a video call is first created (upon
discovery that far-end is video capable) or when
an audio call transitions to a video call (through
far-end transfer), the full screen layout will be
used.
If set to 0 or Null, video calls only use the full
screen layout if it is selected by the user.
video.forceRtcpVideoCodec 0 or 1 0 If set to 1, force the Polycom VVX 1500 to send

Control RTCP feedback messages to request fast update
I-frames for all video calls.
These attributes also include:
• Video Coding Algorithms <codecs/>
• Camera Controls <camera/>
• Local Camera View <localCameraView/>
Video Coding Algorithms <codecs/>

These codecs include:
• Codec Preferences <codecPref/>
• Codec Profiles <profile/>
A - 65
Codec Preferences <codecPref/>
Permitted
video.codecPref.H261 1 to 4 4 Specifies the video codec preferences for

the Polycom VVX 1500 phone.
video.codecPref.H264 1 to 4 1
Note Codecs with a default of Null are available for test purposes only and are not
expected to be used in your deployment.
Codec Profiles <profile/>

The profile attributes can be adjusted for each of the new supported video
codecs.
Permitted
video.profile.H261.jitterBufferMax (video.profil The largest jitter buffer depth to be supported

e.H261.jitter (in milliseconds). Jitter above this size will
BufferMin + always cause lost packets. This parameter
500ms) to should be set to the smallest possible value
2500ms, that will support the expected network jitter.
default 2000ms
video.profile.H261.jitterBufferMin 33ms to The smallest jitter buffer depth (in milliseconds)

1000ms, that must be achieved before play out begins
default 150ms for the first time. Once this depth has been
term average jitter.
video.profile.H261.jitterBufferShrink 33ms to The absolute minimum duration time (in

1000ms, milliseconds) of RTP packet Rx with no packet
default 70ms loss between jitter buffer size shrinks. Use
on known good networks. Use larger values
(1000ms) to minimize packet loss on networks
with large jitter (3000 ms).
A - 66
Configuration Files
Permitted
video.profile.H261.CifMpi 1 (default) to 32 This value is H261 format parameter CIF used

to signal Polycom VVX 1500 receiving
capability in SDP.
This value also controls the TX frame size. If
set to 1, CIF is used (provided the far end
supports CIF=1); otherwise QCIF is used.
video.profile.H261.QcifMpi 1 (default) to 32 This value is H261 format parameter QCIF

used to signal Polycom VVX 1500 receiving
capability in the SDP.
video.profile.H261.annexD 0 or 1 This value is H261 format parameter ANNEXD

default Null used to signal Polycom VVX 1500 receiving
capability in the SDP.

default 2000ms


video.profile.H264.payloadType 96 to 127, RTP payload format type for H264/90000

default 109 MIME type.
video.profile.H264.profileLevel 1, 1b, 1.1, 1.2, This value is H.264's level used in the phone.
1.3 (default) The Level is a constraint set to selected key
algorithm parameters, codec in different level
has different ability, at this time Polycom VVX
1500 support these level (1,1b,1.1,1.2,1.3), as
to detailed level definition. For more
information, refer to ITU-T H.264.
A - 67
Permitted

default 2000ms


video.profile.H263.CifMpi 1 (default) to 32 This value is H263/90000 format parameter

CIF used to signal Polycom VVX 1500
receiving capability in SDP.
video.profile.H263.QcifMpi 1 (default) to 32 This value is H263/90000 format parameter

QCIF used to signal Polycom VVX 1500
receiving capability in the SDP.
video.profile.H263.SqcifMpi 1 (default) to 32 This value is H263/90000 format parameter

SQCIF used to signal Polycom VVX 1500
receiving capability in the SDP.

e.H2631998.ji (in milliseconds). Jitter above this size will
tterBufferMin always cause lost packets. This parameter
+ 500ms) to should be set to the smallest possible value
default 2000ms
A - 68
Configuration Files
Permitted


video.profile.H2631998.payloadType 96 (default) to RTP payload format type for H263-1998/90000

127 MIME type.
video.profile.H2631998.CifMpi 1 (default) to 32 This value is H263-1998/90000 format

parameter CIF used to signal
Polycom VVX 1500 receiving capability in SDP.
video.profile.H2631998.QcifMpi 1 (default) to 32 This value is H263-1998/90000 format

parameter QCIF used to signal
Polycom VVX 1500 receiving capability in the
SDP.
video.profile.H2631998.SqcifMpi 1 (default) to 32 This value is H263-1998/90000 format

parameter SQCIF used to signal
SDP.
video.profile.H2631998.annexF 0 or 1 This value is H263-1998/90000 format

default Null parameter ANNEXF used to signal
SDP.
video.profile.H2631998.annexI 0 or 1 This value is H263-1998/90000 format

default Null parameter ANNEXI used to signal
SDP.
video.profile.H2631998.annexJ 0 or 1 This value is H263-1998/90000 format

default Null parameter ANNEXJ used to signal
SDP.
A - 69
Permitted
video.profile.H2631998.annexT 0 or 1 This value is H263-1998/90000 format

default Null parameter ANNEXT used to signal
SDP.
video.profile.H2631998.annexK 0 or 1 (default), This value is H263-1998/90000 format

2, 3, 4 parameter ANNEXK used to signal
SDP.
video.profile.H2631998.annexN 0 or 1 (default), This value is H263-1998/90000 format

2, 3, 4 parameter ANNEXN used to signal
SDP.
Camera Controls <camera/>

These settings control the performance of the camera.
Permitted
video.camera. 0 to 2 Null Set flicker avoidance.

flickerAvoidance If set to 0 or Null, flicker avoidance is automatic.
If set to 1, 50hz AC power frequency flicker
avoidance (Europe/Asia).
If set to 2, 60hz AC power frequency flicker
avoidance (North America).
video.camera.frameRate 5 to 30 frames Null Set target frame rate.

per second Values indicate a fixed frame rate, from 5 (least
smooth) to 30 (most smooth).
video.camera.brightness 0 to 6 Null Set brightness level.

The value range is from 0 (Dimmest) to 6
(Brightest).
video.camera.saturation 0 to 6 Null Set saturation level.

The value range is from 0 (Lowest) to 6 (Highest).
A - 70
Configuration Files
Permitted
video.camera.contrast 0 to 4 Null Set contrast level.

The value range is from 0 (No contrast increase) to
3 (Most contrast increase, and 4 (Noise reduction
contrast).
video.camera.sharpness 0 to 6 Null Set sharpness level.

The value range is from 0 (Lowest) to 6 (Highest).
Local Camera View <localCameraView/>

These settings control how the local camera is viewed on the screen.
Permitted
video.localCameraView. 0=Disable, Null Determines whether the local camera view is

fullscreen.enabled 1=Enable shown in the full screen layout .
If set to 0, the local camera view is not shown.
If set to 1 or Null, the local camera view is shown.
video.localCameraView. “pip” or Null How the local camera view is shown.

fullscreen.mode Null If set to “pip”, the local camera view appears as a
picture-in-picture with the far end window.
If set to Null, the local camera view appears
side-by-side with the far end window.
Quality of Service <QOS/>

These settings control the Quality of Service (QOS) options.
• Ethernet IEEE 802.1p/Q <ethernet/>
• IP TOS <IP/>
Ethernet IEEE 802.1p/Q <ethernet/>

The following settings control the 802.1p/Q user_priority field:
• RTP <RTP/>
• Call Control <callControl/>
A - 71
• Other <other/>
RTP <RTP/>
These parameters apply to RTP packets.
Permitted
qos.ethernet.rtp.user_priority 0-7 5 User-priority used for Voice RTP

packets.
qos.ethernet.rtp.video.user_priority 0-7 5 User-priority used for Video RTP

packets.
Call Control <callControl/>

These parameters apply to call control packets, such as the network protocol
signaling.
Permitted
qos.ethernet.callControl.user_priority 0-7 5 User-priority used for call control

packets.
Other <other/>
These default parameter values are used for all packets which are not set
explicitly.
Permitted
qos.ethernet.other.user_priority 0-7 2 User-priority used for packets that

do not have a per-protocol setting.
IP TOS <IP/>
The following settings control the “type of service” field in outgoing packets:
• RTP <rtp/>
• Call Control <callControl/>
A - 72
Configuration Files
RTP <rtp/>
These parameters apply to RTP packets.
Permitted
qos.ip.rtp.dscp 0 to 63 or Null This parameter allows the DSCP of

EF or packets to be specified. If set to a
any of value, this will override the other
AF11,AF12, qos.ip.rtp… parameters. Default
AF13,AF21, of Null which means the other
AF22,AF23, qos.ip.rtp… parameters will be
AF31,AF32, used.
AF33,AF41,
AF42,AF43
qos.ip.rtp.min_delay 0 or 1 1 If set to 1, set min-delay bit in the IP

TOS field of the IP header, or else
don’t set it.
qos.ip.rtp.max_throughput 0 or 1 1 If set to 1, set max-throughput bit in

the IP TOS field of the IP header, or
else don’t set it.
qos.ip.rtp.max_reliability 0 or 1 0 If set to 1, set max-reliability bit in

qos.ip.rtp.min_cost 0 or 1 0 If set to 1, set min-cost bit in the IP

don’t set it.
qos.ip.rtp.precedence 0-7 5 If set to 1, set precedence bits in the

IP TOS field of the IP header, or
else don’t set them.
qos.ip.rtp.video.dscp 0 to 63 or Null This parameter allows the DSCP of

any of value, this will override the other
AF11,AF12, qos.ip.rtp.video… parameters.
AF13,AF21, Default of Null which means the
AF22,AF23, other qos.ip.rtp.video…
AF31,AF32, parameters will be used.
AF33,AF41,
AF42,AF43
qos.ip.rtp.video.min_delay 0 or 1 1 If set to 1, set min-delay bit in the IP

don’t set it.
qos.ip.rtp.video.max_throughput 0 or 1 1 If set to 1, set max-throughput bit in

A - 73
Permitted
qos.ip.rtp.video.max_reliability 0 or 1 0 If set to 1, set max-reliability bit in

qos.ip.rtp.video.min_cost 0 or 1 0 If set to 1, set min-cost bit in the IP

don’t set it.
qos.ip.rtp.video.precedence 0-7 5 If set to 1, set precedence bits in the

Call Control <callControl/>

These parameters apply to call control packets, such as the network protocol
signaling.
Permitted
qos.ip.callControl.dscp 0 to 63 or Null This parameter allows the DSCP of

any of value this will override the other
AF11,AF12, qos.ip.callControl…
AF13,AF21, parameters. Default of Null which
AF22,AF23, means the other
AF31,AF32, qos.ip.callControl…
AF33,AF41, parameters will be used.
AF42,AF43
qos.ip.callControl.min_delay 0 or 1 1 If set to 1, set min-delay bit in the IP

don’t set it.
qos.ip.callControl.max_throughput 0 or 1 0 If set to 1, set max-throughput bit in

qos.ip.callControl.max_reliability 0 or 1 0 If set to 1, set max-reliability bit in

qos.ip.callControl.min_cost 0 or 1 0 If set to 1, set min-cost bit in the IP

don’t set it.
qos.ip.callControl.precedence 0-7 5 If set to 1, set precedence bits in the

A - 74
Configuration Files
Basic TCP/IP <TCP_IP/>

• Network Monitoring <netMon/>
• Time Synchronization <sntp/>
• Port <port/>
• Keep-Alive <keepalive/>
Network Monitoring <netMon/>
Permitted
Attribute Values Default
tcpIpApp.netMon.enabled 0 or 1 1
tcpIpApp.netMon.period 1 to 86400 30
Time Synchronization <sntp/>

The following table describes the parameters used to set up time
synchronization and daylight savings time. The defaults shown will enable
daylight savings time (DST) for North America.
Daylight savings defaults:
• Do not use fixed day, use first or last day of week in the month.
• Start DST on the second Sunday in March at 2 am.
• Stop DST on the first Sunday in November at 2 am.
Permitted
tcpIpApp.sntp.resyncPeriod positive 86400 (24 Time in seconds between

integer hours) Simple Network Time
Protocol (SNTP) re-syncs.
tcpIpApp.sntp.address valid host clock Address of the SNTP

name or IP server.
address
A - 75
Permitted
tcpIpApp.sntp.address.overrideDHCP 0 or 1 0 These parameters

determine whether
configuration file
parameters override DHCP
parameters for the SNTP
server address and
Greenwich Mean Time
(GMT) offset. If set to 0,
DHCP values will override
configuration file
parameters. If set to 1, the
configuration file
parameters will override
DHCP values.
tcpIpApp.sntp.gmtOffset positive or -28800 Offset in seconds of the

negative (Pacific local time zone from GMT.
integer time) 3600 seconds = 1 hour
tcpIpApp.sntp.gmtOffset.overrideDHCP 0 or 1 0 These parameters

determine whether
configuration file
parameters override DHCP
parameters for the SNTP
server address and GMT
offset. If set to 0, DHCP
values will override
configuration file
parameters. If set to 1, the
configuration file
parameters will override
DHCP values.
tcpIpApp.sntp.daylightSavings.enable 0 or 1 1 If set to 1, apply daylight

savings rules to displayed
time.
tcpIpApp.sntp.daylightSavings.fixedDayEnable 0 or 1 0 If set to 0, month, date, and

dayOfWeek are used in
DST date calculation.
If set to 1, then only month
and date are used.
tcpIpApp.sntp.daylightSavings.start.month 1-12 3 (March) Month to start DST.

Mapping: 1=Jan, 2=Feb, ...,
12=Dec
A - 76
Configuration Files
Permitted
tcpIpApp.sntp.daylightSavings.start.date 1-31 8 If fixedDayEnable is set to

1, use as day of the month
to start DST.
If fixedDayEnable is set to
0, us the mapping: 1 = the
first occurrence of a given
day-of-the-week in a month,
8 = the second occurrence
of a given day-of-the-week
in a month, 15 = the third
occurrence of a given
day-of-the-week in a month,
22 = the fourth occurrence
of a given day-of-the-week
in a month
tcpIpApp.sntp.daylightSavings.start.time 0-23 2 Time of day to start DST in

24 hour clock.
Mapping: 2=2 am, 14=2 pm
tcpIpApp.sntp.daylightSavings.start.dayOfWeek 1-7 1 Day of week to apply DST.

Mapping: 1=Sun, 2=Mon,
..., 7=Sat
tcpIpApp.sntp.daylightSavings.start.dayOfWeek. 0 or 1 0 If set to 1 and

lastInMonth fixedDayEnable is set to 0,
DST starts on the last day
(specified by
start.dayOfWeek) of the
week in the month. The
start.date is ignored.
tcpIpApp.sntp.daylightSavings.stop.month 1-12 11 Month to stop DST.
tcpIpApp.sntp.daylightSavings.stop.date 1-31 1 Day of the month to stop

DST.
tcpIpApp.sntp.daylightSavings.stop.time 0-23 2 Time of day to stop DST in

24 hour clock.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek 1-7 1 Day of week to stop DST.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek. 0 or 1 0 If set to 1 and

lastInMonth fixedDayEnable set to 0,
DST stops on the last day
(specified by
stop.dayOfWeek) of the
week in the month. The
stop.date is ignored.
A - 77
Port <port/>
• RTP <rtp/>
RTP <rtp/>
Permitted
tcpIpApp.port.rtp.filterByIp 0 or 1 1 If set to 1, reject RTP packets

arriving from (sent from) a
non-negotiated (through SDP) IP
address.
tcpIpApp.port.rtp.filterByPort 0 or 1 0 If set to 1, reject RTP packets

arriving from (sent from) a
non-negotiated (through SDP)
port.
tcpIpApp.port.rtp.forceSend Null, Null When non-Null, send all RTP

1024-65534 packets to, and expect all RTP
packets to arrive on, the
specified port.
Note: both
tcpIpApp.port.rtp.filterByIp and
tcpIpApp.port.rtp.filterByPort
must be enabled for this to work.
tcpIpApp.port.rtp.mediaPortRangeStart Null, even Null If set to Null, the value 2222 will
integer from be used for the first allocated
1024-65534 RTP port, otherwise, the
specified port will be used. Ports
will be allocated from a pool
starting with the specified port up
to a value of (start-port + 47) for
a voice-only phone or (start-port
+ 95) for a video phone.
A - 78
Configuration Files
Keep-Alive <keepalive/>
Allowing for the configuration of TCP keep-alive on SIP TLS connections, the
phone can detect a failures quickly (in minutes) and attempt to re-register with
the SIP call server (or its redundant pair).
Permitted
tcpIpApp.keepalive.tcp.idleTransmitInterval 10 to 7200 Null After idle x seconds, the

keep-alive message is sent to
the call server.
If set to Null, the default value is
30 seconds.
Note: If this parameter is set to a
value that is out of range, the
default value is used.
tcpIpApp.keepalive.tcp. 5 to 120 Null If no response is received to

noResponseTrasmitInterval keep-alive message, another
keep-alive message is sent to
the call server after x seconds.
If set to Null, the default value to
20 seconds.
Note: If this parameter is set to a
value that is out of range, the
default value is used.
tcpIpApp.keepalive.tcp.sip.tls.enable 0 or 1 0 If set to 1, enable TCP keep-alive

for SIP signaling connections
that use TLS transport.
If set to 0, disable TCP
keep-alive for SIP signaling
connections that use TLS
transport.
Web Server <httpd/>

The phone contains a local web server for user and administrator features.
This can be disabled for applications where it is not needed or where it poses
a security threat. The web server supports both basic and digest
authentication. The authentication user name and password are not
configurable for this release.
A - 79
Permitted
httpd.enabled 0 or 1 1 If set to 1, the HTTP server will be enabled.
httpd.cfg.enabled 0 or 1 1 If set to 1, the HTTP server configuration interface will be

enabled.
httpd.cfg.port 1-65535 80 Port is 80 for HTTP servers. Care should be taken when
choosing an alternate port.
Call Handling Configuration <call/>

Permitted
call.rejectBusyOnDnd 0 or 1 1 If set to 1, reject all incoming calls with the

reason “busy” if do-not-disturb is enabled.
Note: This attribute is ignored when the line is
configured as shared. The reason being that
even though one party has turned on DND, the
other person/people sharing that line do not
necessarily want all calls to that number diverted
away.
Note: If server-based DND is enabled, this
parameter is disabled.
call.enableOnNotRegistered 0 or 1 1 If set to 1, calls will be allowed when the phone is

not successfully registered, otherwise, calls will
not be permitted without a valid registration.
Note: Setting this parameter to 1 can allow
Polycom VVX 1500 phones to make calls using
the H.323 protocol even though an H.323
gatekeeper is not configured.
call.offeringTimeOut positive 60 Time in seconds to allow an incoming call to ring

integer before dropping the call, 0=infinite.
Note: The call diversion, no answer feature will
take precedence over this feature if enabled. For
more information, refer to No Answer
<noanswer/> on page A-144.
call.ringBackTimeOut positive 60 Time in seconds to allow an outgoing call to

integer remain in the ringback state before dropping the
call, 0=infinite.
A - 80
Configuration Files
Permitted
call.dialtoneTimeOut Null, positive 60 Time in seconds to allow the dial tone to be

integer played before dropping the call.
If set to 0, the call is not dropped.
If set to Null, call dropped after 60 seconds.
call.lastCallReturnString string of *69 The string sent to the server when the user
maximum selects the “last call return” action.
length 32
call.callsPerLineKey 1 to 24 OR 34, 24, 8 For the SoundPoint IP 650 and 670, the
1 to 8 OR 4 permitted range is 1 to 34 and the default is 34.
For the SoundPoint IP 550 and 560, the
permitted range is 1 to 24 and the default is 24.
For the SoundPoint IP 32x/33x and 430, the
permitted range is 1 to 8 and the default is 4.
For all other phones, the permitted range is 1 to
8 and the default is 8.
This is the number of calls that may be active or
on hold per line key on the phone.
Note that this may be overridden by the
per-registration attribute of
reg.x.callsPerLineKey. Refer to Registration
<reg/> on page A-134.
call.stickyAutoLineSeize Null, 0, or 1 0 If set to 1, makes the phone use "sticky" line

seize behavior. This will help with features that
need a second call object to work with. The
phone will attempt to initiate a new outgoing call
on the same SIP line that is currently in focus on
the LCD (this was the behavior in SIP 1.6.5).
Dialing through the call list when there is no
active call will use the line index for the previous
call. Dialing through the call list when there is an
active call will use the current active call line
index. Dialing through the contact directory will
use the current active call line index.
If set to 0 or Null, the feature is disabled (this was
the behavior in SIP 1.6.6). Dialing through the
call list will use the line index for the previous
call. Dialing through the contact directory will use
a random line index.
Note: This may fail due to glare issues in which
case the phone may select a different available
line for the call.
A - 81
Permitted
call.stickyAutoLineSeize. Null, 0, or 1 Null If call.stickyAutoLineSeize is set to 1, this

onHookDialing parameter has no effect. The regular
stickyAutoLineSeize behavior is followed.
If call.stickyAutoLineSeize is set to 0 or Null
and this parameter is set to 1, this overrides the
stickyAutoLineSeize behavior for hot dial only.
(Any new call scenario seizes the next available
line.)
If call.stickyAutoLineSeize is set to 0 or Null
and this parameter is set to 0 or Null, there is no
difference between hot dial and new call
scenarios.
Note: A hot dial occurs on the line which is
currently in the call appearance. Any new call
scenario seizes the next available line.
call.singleKeyPressConference 0,1 0 If set to 1, the conference will be setup after a

user presses the Conference soft key or
Conference key the first time. Also, all sound
effects (dial tone, DTMF tone while dialing and
ringing back) are heard by all existing
participants in the conference.
If set to 0 or Null, sound effects are only heard by
conference initiator (old behavior).
Only supported for SoundPoint IP 550, 560,650
and 670 and SoundStation IP 7000. For all
others, set to 0.
call.localConferenceCallHold 0 or 1 0 If set to 0, a hold will happen for all legs when

conference is put on hold. (old behavior).
If set to 1, only the host is out of the conference,
all other parties in conference continue to talk.
(new behavior).
If set to Null, the default value is 0.
Only supported for the SoundPoint IP 550,
560,650 and 670 and the SoundStation IP 7000
(refer to Manage Conferences on page 4-22).
For all others, set to 0.
call.transfer.blindPreferred 0,1 Null If set to 1, the blind transfer is the default mode.
The Normal soft key is available to switch to a
consultative transfer.
If set to 0 or Null, the consultative transfer is the
default mode. The Blind soft key is available to
switch to a blind transfer.
SoundPoint IP 32x/33x only.
A - 82
Configuration Files
Permitted
call.directedCallPickupString star code *97 The star code to initiate a directed call pickup.
Note: The default value supports the
BroadWorks calls server only. You must change
the value if your organization uses a different call
server.
call.directedCallPickupMethod “native” or Null The method the phone will use to perform a
“legacy” directed call pick-up of a BLF resource's inbound
ringing call. “native” indicates the phone will use
a native protocol method (in this case SIP
INVITE with the Replaces header [4]). “legacy”
indicates the phone will use the method specified
in call.directedCallPickupString.
call.parkedCallRetrieveMethod “native” or Null The method the phone will use to retrieve a BLF
“legacy” resource's call which has dialog state confirmed.
“native” indicates the phone will use a native
protocol method (in this case SIP INVITE with
the Replaces header [4]). “legacy” indicates the
phone will use the method specified in
call.parkedCallRetrieveString .
call.parkedCallRetrieveString star code Null The star code used to initiate retrieve of a parked
call.
call.autoAnswer.micMute 0 or 1 1 If set to 1, the microphone is initially muted after

a call is auto-answered.
call.autoAnswer.videoMute 0 or 1 0 If set to 1, video Tx is initially disabled after a call

is auto-answered.
call.autoAnswer.SIP 0 or 1 0 If set to 1, auto-answer is enabled for all SIP

calls.
call.autoAnswer.H323 0 or 1 0 If set to 1, auto-answer is enabled for all H.323

calls.
call.autoAnswer.ringClass positive 4 The ring class (se.rt.x) to use when a call is to

integer be automatically answered using the
auto-answer feature. If set to a ring class with a
type other than “answer” or “ring-answer”, the
setting will be overridden such that a ring type of
“visual” (no ringer) applies.
A - 83
Permitted
call.autoRouting.preference “line” or “line If set to line, calls are placed via the first
“protocol” available line, regardless of its protocol
capabilities. If the first available line has both SIP
and H.323 capabilities, the preferred protocol will
be used preferentially
(call.autoRouting.preferredProtocol).
If set to protocol, the first available line with the
preferred protocol activated is used, if available,
and if not available, the first available line will be
used.
Note: Auto-routing is used when manual routing
selection features are disabled. Refer to User
Preferences <up/> on page A-31.
call.autoRouting. “SIP or SIP If set to SIP, calls are placed via SIP if available,
preferredProtocol “H323” or via H.323 if SIP is not available.
If set to H323, calls are placed via H.323 if
available, or via SIP if H.323 is not available.
• Shared Calls <shared/>
• Hold, Local Reminder <hold/><localReminder/>
Shared Calls <shared/>

Permitted
call.shared.disableDivert 0 or 1 1 If set to 1, disable diversion feature for shared

lines.
Note: This feature is disabled on most call
servers.
call.shared.seizeFailReorder 0 or 1 1 If set to 1, play re-order tone locally on shared

line seize failure.
A - 84
Configuration Files
Permitted
call.shared.oneTouchResume 0 or 1 0 If set to 1, when a shared line has a call on hold

the remote user can press that line and resume
the call. If more than one call is on hold on the
line then the first one will be selected and
resumed automatically.
If set to 0, pressing the shared line will bring up
a list of the calls on that line and the user can
select which call the next action should be
applied to.
Note: This parameter affects the SoundStation
IP 6000 and 7000 phones. For other phones, a
quick press and release of the line key will
resume a call whereas pressing and holding
down the line key will show a list of calls on that
line.
call.shared.exposeAutoHolds 0 or 1 0 If set to 1, on a shared line, when setting up a

conference, a re-INVITE will be sent to the
server.
If set to 0, no re-INVITE will be sent to the
server.
Hold, Local Reminder <hold/><localReminder/>

Permitted
call.hold.localReminder.enabled 0 or 1 0 If set to 1, periodically notify the local

user that calls have been on hold for
an extended period of time.
call.hold.localReminder.period non-negative 60 Time in seconds between subsequent

integer reminders.
call.hold.localReminder.startDelay non-negative 90 Time in seconds to wait before the

integer initial reminder.
Directory <dir/>
• Local Directory <local/>
• Corporate Directory <corp/>
A - 85
Local Directory <local/>

The local directory is stored in either flash memory or RAM on the phone. The
local directory size is limited based on the amount of flash memory in the
phone. (Different phone models have variable flash memory.)
When the volatile storage option is enabled, ensure that a properly configured
provisioning server that allows uploads is available to store a back-up copy of
the directory or its contents will be lost when the phone reboots or loses power.
Permitted
dir.local.volatile.2meg 0 or 1 0 Applies to platforms with 2 Mbytes

of flash memory.
If set to 1, use volatile storage for
phone-resident copy of the directory
to allow for larger size.
dir.local.nonVolatile.maxSize.2meg 1 to 20 20 Applies to platforms with 2 Mbytes

of flash memory. Maximum size in
Kbytes of non-volatile storage that
the directory will be permitted to
consume.
dir.local.volatile.4meg 0 or 1 0 Applies to platforms with 4 Mbytes

of flash memory.
dir.local.nonVolatile.maxSize.4meg 1 to 50 50 Applies to platforms with 4 Mbytes

of flash memory. Maximum size in
Kbytes of non-volatile storage that
the directory will be permitted to
consume.
dir.local.volatile.maxSize 1 to 200 200 When the volatile storage option is

set (refer to see
dir.local.volatile.4meg and
dir.local.volatile.8meg), this
attribute is the maximum size of
contact directory file that the phone
supports. Note that phones with 16
MB RAM support up to 50 Kbytes of
directory file, and phones with more
than 16 MB RAM support up to 200
Kbytes of directory file. When the
value specified for this attribute
exceeds the limit, the limit will be
used as the max. directory size.
A - 86
Configuration Files
Permitted
dir.local.volatile.8meg 0 or 1 0 Attribute applies only to platforms

with 8 Mbytes or more of flash
memory.
dir.local.nonVolatile.maxSize.8meg 1 to 100 100 Attribute applies only to platforms

with 8 Mbytes or more of flash
memory.
This is the maximum size of
non-volatile storage that the
directory will be permitted to
consume.
dir.local.readonly 0 or 1 1 Specifies whether or not local

contact directory is read only.
If set to 0 or Null, the local contact
directory is editable.
If set to 1, the local contact directory
is read only.
Note: If the local contact directory is
read only, speed dial entry on the
SoundPoint IP 32x/33x is disabled
(enter the speed dial index followed
by “#”).
dir.search.field 0 or 1 Null Specifies how to search the contact

directory. If set to 1, search by
contact’s first name. If set to 0,
search by contact’s last name.
Corporate Directory <corp/>

A portion of the corporate directory is stored in flash memory on the phone.
The size is based on the amount of flash memory in the phone. (Different
phone models have variable flash memory.)
A - 87
Permitted
dir.corp.address dotted-decimal Null The IP address or host name of the

IP address or LDAP server interface to the
host name or corporate directory. For example,
FQDN host.domain.com.
dir.corp.port 0, Null, 1 to 389 (TCP) This parameter is used to specify

65535 636 (TLS) the port to connect to on the server,
if a full URL is not provided.
dir.corp.transport TCP, TLS, Null TCP This parameter is used to specify

whether a TCP or TLS connection is
made with the server, if a full URL is
not provided.
dir.corp.baseDN UTF-8 encoded Null The base domain name is the

string starting point for making queries on
the LDAP server.
dir.corp.user UTF-8 encoded Null The username used to authenticate

string to the LDAP server.
dir.corp.password UTF-8 encoded Null The password used to authenticate

string to the LDAP server.
dir.corp.filterPrefix UTF-8 encoded (objectclas Predefined filter string.

string s=person) If set to Null or invalid,
“(objectclass=person)” is used.
dir.corp.scope “one”, “sub”, “sub” Type of search.

“base” If set to “one”, a search of the level
one below the baseDN is
performed.
If set to “sub” or Null, a recursive
search (of all levels below the
baseDN) is performed.
If set to “base”, a search at the
baseDN level is performed.
dir.corp.attribute.x.name UTF-8 encoded Null The name of the attribute to match

string on the server. Each name must be
unique, however, an LDAP entry
can have multiple attributes with the
same name.
Up to eight attributes can be
configured (x = 1 to 8).
dir.corp.attribute.x.label UTF-8 encoded Null A UTF-8 encoded string that is used

string as the label when data is displayed.
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Configuration Files
Permitted
dir.corp.attribute.x.type first_name, last_name This parameter defines how the

last_name, attribute is interpreted by the phone.
phone_number, Entries can have multiple attributes
SIP_address, of the same type. Type ‘other’ is
H323_address used for display purposes only.
URL, If the user saves the entry to the
other local contact directory on the
phone, first_name, last_name, and
phone_number are copied. The
user can place a call to the
phone_number and SIP_address
from the corporate directory.
dir.corp.attribute.x.sticky 0 or 1 Null If set to 0 or Null, the filter criteria for

this attribute is reset after a reboot.
If set to 1, the filter criteria for this
attribute is retained through a
reboot.
Such attributes are denoted with a
“*” before the label when displayed
on the phone.
dir.corp.attribute.x.filter UTF-8 encoded Null The filter string for this attribute,
string which is edited when searching.
dir.corp.attribute.x.searchable 0 or 1 0 A flag to determine if the attribute is

searchable through quick search.
This flag applies for x = 2 or greater.
If set to 0 or Null, quick search on
this attribute is disabled.
If set to 1, quick search on this
attribute is enabled.
dir.corp.backGroundSync 0 or 1 0 If set to 0 or Null, there will be no

background downloading from the
LDAP server.
If set to 1, there will be background
downloading of data from the LDAP
server.
dir.corp.backGroundSync.period 3600 to 604800 86400 The corporate directory cache is

seconds refreshed after the corporate
directory feature has not been used
for this period of time.
The default period is 24 hours. The
minimum is 1 hour and the
maximum is 7 days.
A - 89
Permitted
dir.corp.viewPersistence 0 or 1 1 If set to 0, the browse position in the

data on the LDAP server and the
attribute filters are reset for
subsequent usage of the corporate
directory.
If set to 1or Null, the browse
position in the data and the attribute
filters are retained for subsequent
usage of the corporate directory.
dir.corp.cacheSize 8 to 256 128 The maximum number of entries

that can be cached locally on the
phone.
dir.corp.pageSize 8 to 64 32 The maximum number of entries

requested from the corporate
directory server with each query.
dir.corp.vlv.allow 0 or 1 0 A flag to determine whether or not

VLV queries can be made if the
LDAP server supports VLV.
If set to 0, VLV queries are disabled.
If set to 1 or Null, VLV queries are
enabled.
Note: If VLV is enabled,
dir.corp.attribute.x.searchab
le is ignored.
dir.corp.vlv.sortOrder list of attributes Null The list of attributes (in the exact
order) to be used by the LDAP
server when indexing. For example,
sn, givenName, telephoneNumber.
A - 90
Configuration Files
Permitted
dir.corp.autoQuerySubmitTimeout 0 to 60 seconds 0 To control if there is a timeout after

the user stops entering characters
in the quick search and, if there is,
how long the timeout is.
If set to 0, there is not (disabled).
dir.corp.sortCtrl 0 or 1 Null Controls how client makes queries

and does it sort entries locally. It
should not be used by users.
If set to 0 or Null, leave sorting as
negotiated between client and
server.
If set to 1, force sorting of queries.
Note: Polycom does not
recommend setting
dir.corp.sortCtrl to 1 as it
causes excessive LDAP queries. It
should be used to diagnose LDAP
servers with sorting problems only.
Presence <pres/>
The parameter pres.reg is the line number used to send SUBSCRIBE. If this
parameter is missing, the phone will use the primary line to send SUBSCRIBE.
Permitted
pres.reg positive 1 Specifies the line/registration

integer number used to send SUBSCRIBE
for presence. Must be a valid
line/registration number. If the
number is not a valid
line/registration number, it is
ignored.
pres.idleSoftkeys 0 or 1 Null If set to Null or 0, the presence idle

soft keys (MyStat and Buddies) do
not appear.
If set to Null or 1, the presence idle
soft keys appear.
Fonts <font/>
These settings control the phone’s ability to dynamically load an external font
file during boot up. Loaded fonts can either overwrite pre-existing fonts
embedded within the software (not recommended) or can extend the phone’s
A - 91
font support for Unicode ranges not already embedded. The font file must be
a Microsoft .fnt file format. The font file name must follow a specific pattern as
described:
• Font filename:
<fontName>_<fontHeightInPixels>_<fontRange>.<fontExtension>
• <fontName> is a free string of characters that typically carries the meaning

of the font. Examples are “fontFixedSize” for a fixed-size font, or
“fontProportionalSize” for a proportional size font.
• <fontHeightInPixels> describes the font height in number of screen pixels.
• <fontRange> describes the Unicode range covered by this font. Since .fnt
are 256 characters based blocks, the <fontRange> is Uxx00_UxxFF (.fnt
file). For more information, refer to Multilingual User Interface on page
4-29.
• <fontExtension> describes the file type. Either .fnt for single 256
characters font .
If it is necessary to overwrite an existing font, use these
<fontName>_<fontHeightInPixels>:
SoundPoint IP 32x, 33x, 430, 450

“fontProp_10” This is the font used widely in the current implementation.
“fontPropSoftkey_10” This is the soft key specific font.
SoundPoint IP 550, 560, 650, and 670
“fontProp_19” This is the font used widely in the current implementation including for
soft keys.
“fontProp_26” This is the font used to display time (but not date).
“fontProp_x” This is a small font used for the CPU/Load/Net utilization graphs, this
is the same as the “fontProp_10” for the SoundPoint IP 500.
If the <fontName>_<fontHeightInPixels> does not match any of the names

above, then the downloaded font will be applied against all fonts defined in
the phone, which means that you may lose the benefit of fonts being calibrated
differently depending on their usage. For example, the font used to display the
time on the SoundPoint IP 650 is a large font, larger than the one used to
display the date, and if you overwrite this default font with a unique font, you
lose this size aspect. For example:
• to overwrite the font used for SoundPoint IP 500 soft keys for ASCII, the
name should be fontPropSoftkey_10_U0000_U00FF.fnt .
• to add support for a new font that will be used everywhere and that is not
currently supported. For example, for the Eastern/Central European
Czech language, this is Unicode range 100-17F, the name could be
fontCzechIP500_10_U0100_U01FF.fnt and
fontCzechIP600_19_U0100_U01FF.fnt .
A - 92
Configuration Files
The font delimiter is important to retrieve the different scrambled .fnt blocks.
This font delimiter must be placed in the “copyright” attribute of the .fnt
header. If you are simply adding or changing a few fonts currently in use,
multiple .fnt files are recommended since they are easier to work with
individually.
font.delimiter string up to 256 ASCII Null Delimiter required to retrieve different

characters grouped .fnt blocks.
• IP_330 font <IP_330/>
• IP_400 font <IP_400/>
• IP_500 font <IP_500/>
• IP_600 font <IP_600/>
IP_330 font <IP_330/>

font.IP_330.x.name fontName_height_Uxx00_U Null Defines the font file that will be loaded from
xxFF.fnt provisioning server during boot up.
Note: When several font.IP_330.x.name
are defined, the index x must follow
consecutive increasing order.
IP_400 font <IP_400/>

A - 93
IP_500 font <IP_500/>

IP_600 font <IP_600/>

font.IP_600.x.name fontName_height_Uxx00 Null Defines the font file that will be loaded from
_UxxFF.fnt provisioning server during boot up.
Keys <key/>
These settings control the scrolling behavior of keys and can be used to change
key functions.
Permitted
key.scrolling.timeout positive 1 The time-out after which a key that is enabled for
integer scrolling will go into scrolling mode until the key is
released. Keys enabled for scrolling are menu
navigation keys (left, right, up, down arrows), volume
keys, and some context-specific soft keys. The value is
an integer multiple of 500 milliseconds (1=500ms).
SoundPoint IP 32x/33x, 430, 450, 550, 560, 650, and 670, SoundStation IP 6000
and 7000, and Polycom VVX 1500 key functions can be changed from the
factory defaults, although this is typically not necessary. For each key whose
function you wish to change, add an XML attribute in the format described in
the following table to the <keys .../> element of the configuration file. These
will override the built-in assignments.
Polycom does not recommend the remapping for keys.
A - 94
Configuration Files
In the following table, x=IP_ IP_330, IP 430, IP_450, IP_550, IP_650, and
IP_6000, and IP_7000 and y is the key number. Note that IP_330 parameters
affect SoundPoint IP 32x/33x phones, IP_430 parameters affect SoundPoint IP
430 phones, IP_550 parameters affect SoundPoint IP 550 and 560 phones,
IP_650 parameters affect SoundPoint IP 650 and 670 phones, IP_6000
parameters affect the SoundStation IP 6000 phones, IP_7000 parameters affect
the SoundStation IP 7000 phones, and the VVX_1500 parameters affect the
Polycom VVX 1500 phones. IP 330: y=1-34; IP 430: y=1-35; IP_450: y=1-35;
IP_550: y=1-40; IP_650:y=1-42; IP_6000:y=1-29; IP_7000:y=1-30;
VVX_1500:y=1-42.
key.x.y.function.prim Functions listed below. Sets the function for key y on platform x.
key.x.y.subPoint.prim positive integer Sets the sub-identifier for key functions with
a secondary array identifier such as
SpeedDial.
The following table lists the functions that are available:
Functions
ArrowDown Dialpad5 Line2 Select
ArrowLeft Dialpad6 Line3 Setup
ArrowRight Dialpad7 Line4 SoftKey1
ArrowUp Dialpad8 Line5 SoftKey2
BuddyStatus Dialpad9 Line6 SoftKey3
CallList DialpadStar Messages SoftKey4
Conference DialpadPound Menu SpeedDial
Delete Directories MicMute SpeedDialMenu
Dialpad0 DoNotDisturb MyStatus Transfer
Dialpad1 Handsfree Null Video
Dialpad2 Headset Offline VolDown
Dialpad3 Hold Redial VolUp
Dialpad4 Line1 Release
Backgrounds <bg/>
The backgrounds used by the SoundPoint IP 450, 550, 560, 650, and 670 and the
Polycom VVX 1500 phones are defined in this section. In the following table,
w=1 to 3, x=1 to 6. hiRes parameters are used by SoundPoint IP 550, 560, 650,
and 670 phones, medRes parameters are used by SoundPoint IP 450 phones,
and VVX_1500 parameters are used by Polycom VVX 1500 phones.
A - 95
Permitted
bg.VVX_1500.color.selection w,x 1,1 Specify which type of background (w) and index
for that type (x) is selected on reboot where w=1
to 3, x=1 to 6.
The type of backgrounds are built-in (w=1), solids
(w=2), and bitmaps (w=3).
w=2 is used when selecting any image as a
background.
w=3 is used when selecting any image from the
Digital Picture Frame as a background. This
image is stored under “Local File”. Only one local
file at a time is supported.
bg.VVX_1500.color.bm.x.name any string Null Graphic files for display on the phone.
For example, if you set
bg.VVX_1500.color.bm.1.name to
Polycom.bmp, the user will be able to select
“Polycom.bmp” as a background on the phone.
bg.hiRes.color.selection w,x 1,1 Specify which type of background (w) and index
for that type (x) is selected on reboot where w=1
to 3, x=1 to 6.
bg.hiRes.color.pat.solid.x. any string Solid pattern name.

name For x=1: Light Blue, x=2: Teal, x=3: Tan, x=4:Null
bg.hiRes.color.pat.solid.x.red 0 to 255 The screen background layouts.
For x=1, red (151), green, (207), blue (249)
bg.hiRes.color.pat.solid.x. 0 to 255
green For x=2, red (73), green (148), blue (148)
For x=3, red (245), green (157), blue (69)
bg.hiRes.color.pat.solid.x.blue 0 to 255
For x=4, red (Null), green (Null), blue (Null)
A - 96
Configuration Files
Permitted
bg.hiRes.color.bm.x.name any string built-in Graphic files for display on the phone and
value of Expansion Module.
bg.hiRes.color.bm.x.em.name any string “Thistle” For x=1:
• name is “Leaf.jpg”
name is “LeafEM.jpg”
For x=2:
• name is “Sailboat.jpg”
name is “SailboatEM.jpg”
For x=3:
• name is “Beach.jpg”
name is “BeachEM.jpg”
For x=4:
• name is “Palm.jpg”
name is “PalmEM.jpg”
For x=5:
• name is “Jellyfish.jpg”
name is “JellyfishEM.jpg”
For x=6:
• name is “Mountain.jpg”
name is “MountainEM.jpg”
Note: If the file is missing or unavailable, the
built-in default solid pattern is displayed.
bg.hiRes.gray.selection w,x 2,1 Specify which type of background (w) and index
(x) for that type is selected on reboot.
bg.hiRes.gray.pr.x.adj -3 Specify the brightness adjustment to the graphic.
bg.hiRes.gray.pat.solid.x.name any string White Solid pattern name.

For x=1: White, x=2: Light Gray, x=3, 4: Null
bg.hiRes.gray.pat.solid.x.red 0 to 255 The screen background layouts.

bg.hiRes.gray.pat.solid.x.green 0 to 255
For x=2, red (160), green (160), blue (160)
bg.hiRes.gray.pat.solid.x.blue 0 to 255 For x=3 and 4, all values are Null.
Note: The values for red, green, and blue must be
the same to display correctly on grayscale.
A - 97
Permitted
bg.hiRes.gray.bm.x.name any string Graphic files for display on the phone and
Expansion Module and also the brightness
bg.hiRes.gray.bm.x.em.name any string adjustment to the graphic.
bg.hiRes.gray.bm.x.adj integer For x=1:
• name is “Leaf.jpg”
name is “LeafEM.jpg”
adjustment is “0”
For x=2:
• name is “Sailboat.jpg”
name is “SailboatEM.jpg”
adjustment is “-3”
For x=3:
• name is “Beach.jpg”
name is “BeachEM.jpg”
For x=4:
• name is “Palm.jpg”
name is “PalmEM.jpg”
For x=5:
• name is “Jellyfish.jpg”
name is “JellyfishEM.jpg”
For x=6:
• name is “Mountain.jpg”
name is “MountainEM.jpg”
Note: The adjustment value is changed on each
individual phone when the user lightens or
darkens the graphic during preview.
bg.medRes.gray.selection w,x 2,1 Specify which type of background (w) and index
(x) for that type is selected on reboot.
bg.medRes.gray.pr.x.adj -3 Specify the brightness adjustment to the graphic.
bg.medRes.gray.pat.solid.x. any string White Solid pattern name.

name For x=1: White, x=2: Light Gray, x=3, 4: Null
bg.medRes.gray.pat.solid.x.red 0 to 255 The screen background layouts.

bg.medRes.gray.pat.solid.x. 0 to 255
green For x=2, red (160), green (160), blue (160)
For x=3 and 4, all values are Null.
bg.medRes.gray.pat.solid.x. 0 to 255
blue Note: The values for red, green, and blue must be
the same to display correctly on grayscale.
A - 98
Configuration Files
Permitted
bg.medRes.gray.bm.x.name any string Graphic files for display on the phone and
Expansion Module and also the brightness
bg.medRes.gray.bm.x.em. any string adjustment to the graphic.
name
For x=1:
bg.medRes.gray.bm.x.adj integer • name is “Leaf256x116.jpg”
For x=2:
• name is “Sailboat256x116.jpg”
For x=3:
• name is “Beach256x116.jpg”
For x=4:
• name is “Palm256x116.jpg”
For x=5:
• name is “Jellyfish256x116.jpg”
For x=6:
• name is “Mountain256x116.jpg”
Note: The adjustment value is changed on each
individual phone when the user lightens or
darkens the graphic during preview.
button.color.selection.x.y. any string The label color for soft keys and line key labels
modify associated with the defined colored backgrounds.
These values can be modified locally by the user.
The format is:
“rgbHILo, <parameter list>”.
For example:
“rbgHiLo, 51, 255, 68, 255, 0, 119” is the default
button color associated with the built-in
background.
button.gray.selection.x.y. any string The label color for soft keys and line key labels
modify associated with the defined gray backgrounds.
These values can be modified locally by the user.
The format is:
“rgbHILo, <parameter list>”.
By default, all defaults are set to “none”.
A - 99
Bitmaps <bitmap/>
The bitmaps used by each phone model are defined in this section.
Platform <IP 330/>, <IP_400/>, <IP_450/>, <IP_600/>, <IP_4000/>, and

<IP_7000/>
In the following table, x=IP_330, IP_400, IP_450, IP_600, IP_4000, or IP_7000
and y is the bitmap number. Note that IP_330 parameters affect SoundPoint IP
32x/33x phones, IP_400 parameters affect SoundPoint IP 430 phones, IP_450
parameters affect SoundPoint IP 450 phones, IP_600 parameters affect
SoundPoint IP 550, 560, 600, 601, and 650, and 670 phones, IP_4000 parameters
affect SoundStation IP 6000 phones, and IP_7000 parameters affect
SoundStation IP 7000 phones.
bitmap.x.y.name The name of a bitmap This is the name of a bitmap to be used for creating an
to be used. animation. If the bitmap is to be downloaded from the
provisioning server, its name must:
• Be different from any name already in use in sip.cfg.
• Match the name of the corresponding <fileName>.bmp to
be retrieved from the provisioning server.
Indicators <ind/>
The following indicators are used by the phone:
• Animations <anim/> <IP_330/>, <IP_335/>, <IP_400/>, <IP_450/>,

<IP_600/>, <IP_4000/>, and <IP_7000/>
• Patterns <pattern/>
• Classes <class/>
• Assignments
A - 100
Configuration Files
Permitted
ind.idleDisplay.mode 1 (default), 2, Null The idle display animation screen layouts.

3 For example, for the SoundPoint IP 330/320:
• If set to 1 or Null, the idle display animation
size is 87 x 11 pixels.
• If set to 2, the idle display animation size is 87
x 22 pixels.
• If set to 3, the idle display animation size is
102 x 22 pixels.
ind.idleDisplay.enabled 0 or 1 0 If set to 1, the idle display may support

presentation of a custom animation if configured
in the animation section of sip.cfg.
Animations <anim/> <IP_330/>, <IP_335/>, <IP_400/>, <IP_450/>,

<IP_600/>, <IP_4000/>, and <IP_7000/>
This section defines bitmap animations composed of bitmap/duration
couples. In the following table, x=IP_330, IP_335, IP_400, IP_450, IP_600,
IP_4000 or IP_7000, y is the animation number, z is the step in the animation.
Note that IP_330 parameters affect SoundPoint IP 32x/33x phones, IP_335
parameters affect SoundPoint IP 335 phones, IP_400 parameters affect
SoundPoint IP 430 phones, IP_450 parameters affect SoundPoint IP 450
phones, IP_600 parameters affect SoundPoint IP 550, 560, 600, 601, 650, and 670
phones, IP_4000 parameters affect SoundStation IP 6000 phones, and IP_7000
parameters affect SoundStation IP 7000 phones.
Note As of SIP 2.2.0, a maximum of 24 frames per animation is supported.
ind.anim.x.y.frame.z.bitmap A bitmap name defined Bitmap to use.

previously. Note that it must be defined already, refer to
Platform <IP 330/>, <IP_400/>, <IP_450/>,
<IP_600/>, <IP_4000/>, and <IP_7000/> on
page A-100.
ind.anim.x.y.frame.z.duration positive integer Duration in milliseconds for this step. 0=infinite.
A - 101
Patterns <pattern/>
This section defines patterns for the LED indicators. In the following table, x is
the pattern number, y is the step in the pattern.
Permitted
ind.pattern.x.step.y.state On or Off Turn LED on or off for this step.
ind.pattern.x.step.y.duration positive integer Duration in milliseconds for this step. 0=infinite
ind.pattern.x.step.y.colour Red or Green For bi-color LEDs, specify color.

(default is Red if
not specified)
Classes <class/>
This section defines the available classes for the LED and graphical icon
indicator types. In the following table, x is the class number, y is the identifier
of the state number for that class.
Permitted
ind.class.x.state.y.index positive integer For LED type indicators, index refers to the pattern index,
such as index x in the Patterns <pattern/> tag above.
For Graphic Icon type indicators, index refers to the
animation index, such as index y in the Animations <anim/>
<IP_330/>, <IP_335/>, <IP_400/>, <IP_450/>, <IP_600/>,
<IP_4000/>, and <IP_7000/> tag above.
Assignments
This attribute assigns a type and a class to an indicator. In the case of the
Graphic Icon type, it also assigns a physical location and size in pixels on the
LCD display (refer to the next section). In the case of the LED type, it assigns a
physical LED number (refer to Graphic Icons <gi/> <IP_330>, <IP_400/>,
<IP_450/>, <IP_600/>, <IP_4000/>, and <IP_7000/> on page A-103).
A - 102
Configuration Files
LEDs <led/>
In the following table, x is the LED number.
Permitted
ind.led.x.index This is for internal usage only and should not be changed (this is
the logical index).
ind.led.x.class positive integer Assigns the class (defined in Classes <class/> on page A-102) for
this indicator.
ind.led.x.physNum This maps the logical index to a specific physical LED.
Graphic Icons <gi/> <IP_330>, <IP_400/>, <IP_450/>, <IP_600/>, <IP_4000/>,

and <IP_7000/>
In the following table, x=IP_330, IP_400, IP_500, IP_600, IP_4000, or IP_7000, y
is the graphic icon number. Note that IP_330 parameters affect SoundPoint IP
32x/33x phones, IP_400 parameters affect SoundPoint IP 430 phones, IP_450
parameters affect SoundPoint IP 450 phones, and IP_600 parameters affect
SoundPoint IP 550, 560, 600, 601, 650, and 670 phones, IP_4000 parameters
affect SoundStation IP 6000 phones, and IP_7000 parameters affect
SoundStation IP 7000 phones.
Permitted
ind.gi.x.y.index This is for internal usage only and should not be changed (this is
the logical index).
ind.gi.x.y.class positive integer Assigns the class (defined in Classes <class/> on page A-102) for
this indicator.
ind.gi.x.y.physX IP 330: 0-101 For Graphic Icon type indicators, this is the x-axis location of the
IP 400: 0-122 upper left corner of the indictor measured in pixels from left to
right.
IP 450: 0-238
IP 600: 0-319
IP 4000: 0-247
IP 7000: 0-255
A - 103
Permitted
ind.gi.x.y.physY IP 330: 0-19 For Graphic Icon type indicators, this is the y-axis location of the
IP 400: 0-45 upper left corner of the indicator measured in pixels from top to
bottom.
IP 450: 0-89
IP 600: 0-159
IP 4000: 0-67
IP 7000: 0-127
ind.gi.x.y.physW IP 330: 1-87 For Graphic Icon type indicators, this is the width of the indicator
IP 400: 1-102 measured in pixels.
IP 450: 1-170
IP 600: 1-320
IP 4000: 1-248
IP 7000: 1-256
ind.gi.x.y.physH IP 330: 1-20 For Graphic Icon type indicators, this is the height of the indicator
IP 400: 1-23 measured in pixels.
IP 450: 1-73
IP 600: 1-160
IP 4000: 1-68
IP 7000: 1-128
Event Logging <log/>
Warning Logging parameter changes can impair system operation. Do not change any
logging parameters without prior consultation with Polycom Technical Support.
The event logging system supports the following classes of events:
Level Interpretation
0 Debug only
1 High detail event class
2 Moderate detail event class
3 Low detail event class
4 Minor error - graceful recovery
5 Major error - will eventually incapacitate the system
6 Fatal error
A - 104
Configuration Files
Each event in the log contains the following fields separated by the | character:
• time or time/date stamp
• 1-5 character component identifier (such as “so”)
• event class
• cumulative log events missed due to excessive CPU load
• free form text - the event description
Example:
011511.006|so |2|00|soCoreAudioTermChg: chassis -> idle
time stamp
ID
event class
missed events
text
Three formats are available for the event timestamp:
Type Example
0 - seconds.milliseconds 011511.006 -- 1 hour, 15 minutes, 11.006 seconds since

booting.
1 - absolute time with minute resolution 0210281716 -- 2002 October 28, 17:16
2 - absolute time with seconds resolution 1028171642 -- October 28, 17:16:42
Two types of logging are supported:
• Basic Logging <level/><change/> and <render/>
• Scheduled Logging Parameters <sched/>
A - 105
Basic Logging <level/><change/> and <render/>

Permitted
log.level.change.xxx 0-5 4 Control the logging detail level for

individual components. These are
the input filters into the internal
memory-based log system.
Possible values for xxx are so,
app1, sip, sspsc, ssps, pps, net,
cfg, cdp, pmt, ftp, ares, dns, cxss,
httpd, rdisk, copy, slog, res, key,
log, curl, rtos, mb, ib, sotet, ttrs,
srtp, usb, efk, clink, ldap, and peer,
pnetm, cmp, cmr, usbio, pres,
pwrsv, and lldp, wmgr, push, poll,
and h323.
log.render.level 0-6 1 Specifies the lowest class of event

that will be rendered to the log files.
This is the output filter from the
internal memory-based log system.
The log.render.level maps to
syslog severity as follows:
0 -> SeverityDebug (7)
1 -> SeverityDebug (7)
2 -> SeverityInformational (6)
3 -> SeverityInformational (6)
4 -> SeverityError (3)
5 -> SeverityCritical (2)
6 -> SeverityEmergency (0)
7 -> SeverityNotice (5)
Syslog Menu on page 3-13.
log.render.type 0-2 2 Refer to above table for timestamp

type.
log.render.realtime 0 or 1 1 Set to 1.
Note: Polycom recommends that
you do not change this value.
log.render.stdout 0 or 1 1 Set to 1.
log.render.file 0 or 1 1 Set to 1.
A - 106
Configuration Files
Permitted
log.render.file.size positive 16 Maximum local application log file

integer, 1 to size in Kbytes. When this size is
179.5 exceeded, the file is uploaded to
the provisioning server and the
local copy is erased.
log.render.file.upload.period positive 172800 Time in seconds between log file

integer uploads to the provisioning server.
Note: The log file will not be
uploaded if no new events have
been logged since the last upload.
log.render.file.upload.append 0 or 1 1 If set to 1, use append mode when

uploading log files to server.
Note: HTTP and TFTP don’t
support append mode unless the
server is set up for this.
log.render.file.upload.append.sizeLimit positive 512 Maximum log file size on

integer provisioning server in Kbytes.
log.render.file.upload.append.limitMode delete, stop delete Behavior when server log file has
reached its limit.
delete=delete file and start over
stop=stop appending to file
Scheduled Logging Parameters <sched/>

The phone can be configured to schedule certain advanced logging tasks on a
periodic basis. These attributes should be set in consultation with Polycom
Technical Support. Each scheduled log task is controlled by a unique attribute
set starting with log.sched.x where x identifies the task.
Permitted
log.sched.x.name alphanumeric Name of an internal system command to be periodically executed.

string To be supplied by Polycom.
log.sched.x.level 0-5 Event class to assign to the log events generated by this command.
This needs to be the same or higher than log.level.change.slog for
these events to appear in the log.
log.sched.x.period positive Seconds between each command execution. 0=run once

integer
A - 107
Permitted
log.sched.x.startMode abs, rel Start at absolute time or relative to boot.
log.sched.x.startTime positive Seconds since boot when startMode is rel or the start time in 24-hour
integer OR clock format when startMode is abs.
hh:mm
log.sched.x.startDay 1-7 When startMode is abs, specifies the day of the week to start
command execution. 1=Sun, 2=Mon, ..., 7=Sat
Security <sec/>
This attribute’s settings affect security aspects of the phone.
.
Permitted
sec.tagSerialNo 0 or 1 Null If set to 1, the phone may advertise its serial number
(Ethernet address) through protocol signaling.
If set to 0 or Null, the phone does advertise its serial
number.
• Encryption <encryption/>
• Password Lengths <pwd/><length/>
• SRTP <srtp/>
• H.235 <H235/>
A - 108
Configuration Files
Encryption <encryption/>
Permitted
sec.encryption.upload.dir 0 or 1 0 If set to 0, the phone-specific contact directory is

uploaded to the server unencrypted regardless of
how it was downloaded. This will replace whatever
phone-specific contact directory is on the server
even if it is encrypted.
If set to 1, the phone-specific contact directory is
uploaded encrypted regardless of how it was
downloaded. This will replace whatever
phone-specific contact directory is on the server
even if it is unencrypted.
sec.encryption.upload. 0 or 1 0 If set to 0, the phone-specific configuration override

overrides file (<Ethernet Address>-phone.cfg) is uploaded
unencrypted regardless of how it was downloaded.
This will replace the override file on the server
even if it is encrypted.
If set to 1, the phone-specific configuration override
file is uploaded encrypted regardless of how it was
downloaded. This will replace the override file on
the server even if it is unencrypted.
Password Lengths <pwd/><length/>

Permitted
sec.pwd.length.admin 0-32 1 Password changes will need to be at least this

long. Use 0 to allow null passwords.
sec.pwd.length.user 0-32 2
A - 109
SRTP <srtp/>
Note As per RFC 3711, you cannot turn off authentication of RTCP.

.
Permitted
sec.srtp.enable 0 or 1 Null If set to 1 or Null, the phone accepts SRTP offers.

If set to 0, the phone always declines SRTP offers.
Note: The default behavior changed in SIP 3.2.0 .
In previous SIP releases, the default value was 0
when null or not defined.
sec.srtp.offer 0 or 1 Null If set to 1 or Null, the phone includes a secure

media stream description along with the usual
non-secure media description in the SDP of a SIP
INVITE. This is for the phone initiating (offering) a
phone call.
If set to 0, no secure media stream is included in
SDP of a SIP invite.
sec.srtp.require 0 or 1 Null If set to 1, the phone is only allowed to use secure

media streams. Any offered SIP INVITEs must
include a secure media description in the SDP or
the call will be rejected. For outgoing calls, only a
secure media stream description is include in the
SDP of the SIP INVITE, meaning that the
non-secure media description is not included. If
sec.srtp.require is set to 1, sec.srtp.offer is
logically set to 1 no matter what the value in the
configuration file.
If set to 0 or Null, secure media streams are not
required.
sec.srtp.offer. 0 or 1 Null If set to 1 or Null, a crypto line with the

HMAC_SHA1_80 AES_CM_128_HMAC_SHA1_80 crypto-suite will
be included in offered SDP.
If set to 0, the crypto line is not included.
Note: This parameter was added in SIP 2.2.1 .
sec.srtp.offer. 0 or 1 Null If set to 1, a crypto line with the

HMAC_SHA1_32 AES_CM_128_HMAC_SHA1_32 crypto-suite will
be included in offered SDP.
If set to 0 or Null, the crypto line is not included.
Note: This parameter was added in SIP 2.2.1 .
A - 110
Configuration Files
Permitted
sec.srtp.key.lifetime 0, positive Null The master key lifetime used for the cryptographic
integer attribute in the SDP. The value specified is the
minimum 1024 number of SRTP packets.
If set to 0 or Null, the master key lifetime is not set.
If set to 1 or greater, master key lifetime is set.
The default setting should be suitable for most
installations. When the lifetime is set greater than 0,
a re-invite with a new key will be sent when the
number of SRTP packets sent for an outgoing call
exceeds half the value of the master key lifetime.
Note: Setting this parameter to a non-zero value
may affect performance of the phone.
sec.srtp.mki.enabled 0 or 1 Null The master key identifier (MKI) is an optional

parameter for the cryptographic attribute in the SDP
that uniquely identifies the SRTP stream within an
SRTP session. MKI is expressed as a pair of
decimal numbers in the form: |mki:mki_length|
where mki is the MKI value and mki_length its
length in bytes.
If set to 1, a four-byte MKI parameter is sent within
the SDP message of the SIP INVITE / 200 OK.
If set to 0 or Null, the MKI parameter is not sent.
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no authentication of RTP is offered. A

noAuth.offer session description that includes the
UNAUTHENTICATED_SRTP session parameter is
sent when initiating a call.
If set to 0 or Null, authentication is offered.
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no authentication of RTP is required.

noAuth.require A call placed to a phone configured with
noAuth.require must offer the
UNAUTHENTICATED_SRTP session parameter in
its SDP.
If sec.srtp.sessionParams.noAuth.require is
set to 1, sec.srtp.sessionParams.noAuth.offer
is logically set to 1 no matter what the value in the
configuration file.
If set to 0 or Null, authentication is required.
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no encryption of RTCP is offered. A

noEncrypRTCP.offer session description that includes the
UNENCRYPTED_SRTCP session parameter is
sent when initiating a call.
If set to 0, or Null, encryption of RTCP is offered.
A - 111
Permitted
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no encryption of RTCP is required.

noEncrypRTCP.require A call placed to a phone configured with
UNENCRYPTED_SRTCP session parameter in its
SDP.
If
sec.srtp.sessionParams.noEncryptRTCP.requi
re is set to 1,
sec.srtp.sessionParams.noEncryptRTCP.offer
configuration file.
If set to 0 or Null, encryption of RTCP is required.
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no encryption of RTP is offered. A

noEncrypRTP.offer session description that includes the
UNENCRYPTED_SRTP session parameter is sent
when initiating a call.
If set to 0, or Null, encryption of RTP is offered.
sec.srtp.sessionParams. 0 or 1 Null If set to 1, no encryption of RTP is required.

noEncrypRTP.require A call placed to a phone configured with
UNENCRYPTED_SRTP session parameter in its
SDP.
If
sec.srtp.sessionParams.noEncryptRTP.requir
e is set to 1,
sec.srtp.sessionParams.noEncryptRTP.offer
configuration file.
If set to 0 or Null, encryption of RTP is required.
sec.srtp.requireMatchingTag 0 or 1 Null A flag to determine whether or not to check the tag

value in the crypto attribute in an SDP answer.
If set to 1 or Null, the tag values must match.
If set to 0, the tag value is ignored.
H.235 <H235/>
Note At this time, this attribute is used with the Polycom VVX 1500 phone only.
The H.235 Voice Profile implementation is Polycom HDX-compatible.
OpenSSL-based Diffie-Hellman key exchange and AES-128 CBC encryption
algorithms is used to encrypt the RTP media
A - 112
Configuration Files
Permitted
sec.H235. 0 or 1 1 If set to 1, H.235 Voice Profile RTP media

mediaEncryption. encryption will be enabled. When enabled, media
enabled encryption will be negotiated when such encryption
is requested by the far end.
sec.H235. 0 or 1 0 If set to 1 and

mediaEncryption.offer sec.H235.mediaEncryption.enabled is also set
to 1, media encryption negotiations will be initiated
with the far end;however, successful negotiations
is not a requirement for the call to complete.
sec.H235. 0 or 1 0 If set to 1 and

mediaEncryption.require sec.H235.mediaEncryption.enabled is also set
to 1, media encryption negotiations will be initiated
or completed with the far end, and if negotiations
fail, the call will be dropped.
License <license/>
This attribute’s settings control aspects of the feature licensing system.
Permitted
license.polling.time 00:00 – 23:59 2:00am The time to check whether or not the license has
expired.
A - 113
Provisioning <prov/>
This attribute’s settings control aspects of the phone’s provisioning server
provisioning system.
Permitted
prov.fileSystem.rfs0.minFreeSpace 5-512 5 Minimum free space in Kbytes to

reserve in the file system when
prov.fileSystem.ffs0.4meg.minFreeSpace 420 downloading files from the
prov.fileSystem.ffs0.2meg.minFreeSpace 48
prov.fileSystem.ffs0.8meg.minFreeSpace 512 you do not change these
parameters.
Note: For the SoundPoint IP 650
phone,
prov.fileSystem.ffs0.8meg.m
inFreeSpace is internally
replaced by 2X the value.
Note: For the SoundPoint IP 7000
phone,
prov.fileSystem.rfs0.minFre
eSpace is internally replaced by
4X the value.
prov.polling.enabled 0 or 1 0 If set to 1, automatic periodic

provisioning server polling for
upgrades is enabled.
prov.polling.mode abs, rel abs Polling mode is absolute or

relative.
prov.polling.period integer 86400 Polling period in seconds.

greater than Rounded up to the nearest
3600 number of days in abs mode.
Measured relative to boot time in
rel mode.
prov.polling.time Format is 03:00 Only used in abs mode. Polling

hh:mm time.
prov.quickSetup.enabled 0 or 1 Null If set to 1, the quick setup feature

is enabled.
If set to 0 or Null, the quick setup
feature is disabled.
A - 114
Configuration Files
RAM Disk <ramdisk/>

This attribute’s settings control the phone’s internal RAM disk feature.
Permitted
ramdisk.enable 0 or 1 1 If set to 1, RAM disk will be available. The RAM

disk is used to cache downloaded wave files, and
other resources for the user interface.
ramdisk.bytesPerBlock 0, 32, 33, ..., 0 These four parameters use internal defaults when
1024 value is set to 0.
Note: For the SoundPoint IP 650 phone,
ramdisk.blocksPerTrack 0, 1, 2, ..., 0
ramdisk.bytesPerBlock is internally replaced by 2X
65536
the value.
ramdisk.nBlocks 0, 1, 2, ..., 4096 Note: For the SoundPoint IP 7000 phone,
65536 ramdisk.bytesPerBlock is internally replaced by 4X
the value.
ramdisk.nBlocks.IP_650 0, 1, 2, ..., 2048
65536
ramdisk.minsize 50 to 16384 50 Smallest size in Kbytes of RAM disk to create

before returning an error. RAM disk size is variable
depending on the amount of device memory.
ramdisk.minfree 512 to 16384 3150 Minimum amount of free space that must be left
after the RAM disk has been created. The RAM
disk’s size will be reduced as necessary in order to
leave this amount of free RAM.
Request <request/>
• Delay <delay/>
A - 115
Delay <delay/>
These settings control the phone’s behavior when a request for restart or
reconfiguration is received.
Permitted
request.delay.type Null, “audio”, or call Defines the strategy to adopt before a request gets
“call” executed. If set to “audio”, a request can be executed as
soon as there is no active audio on the phone,
independently of any call state. If set to “call”, a request
can be executed as soon as there are no calls in any
state on the phone.
Feature <feature/>
These settings control the activation or deactivation of a feature at run time. In
the table below, x is the feature number.
feature.x.name “presence” “presence” is the presence feature including management of

buddies and own status
“messaging” “messaging” is the instant messaging feature
“directory” “directory” is the local directory feature
“calllist” “calllist” is the locally controlled call lists

Note: The “call list” feature can be disabled on all
SoundPoint IP, SoundStation IP, and VVX phones except the
SoundPoint IP 32x/33x.
“ring-download” “ring-download” is run-time downloading of ringers
“calllist-received” “calllist-received” is the received-calls list feature (the

“calllist” feature must be enabled for this feature to be
available)
“calllist-placed” “calllist-placed” is the placed-calls list feature (the “calllist”

feature must be enabled for this feature to be available)
“calllist-missed” “calllist-missed” is the missed-calls list feature (the “calllist”

feature must be enabled for this feature to be available)
“url-dialing” “url-dialing” controls whether URL/name dialing is available

from a private line (it is never available from a shared line)
Note: The "url-dialing" feature must be disabled by setting
feature.9.enabled to 0 in order to prevent unknown
callers from being identified on the display by an IP address.
“call-park” “call-park” is the call park and park-retrieve features
A - 116
Configuration Files
feature.x.name “group-call-pickup” “group-call-pickup” is the group call pickup feature

(continued)
“directed-call-pickup” “directed-call-pickup” is the directed call pickup feature
“last-call-return” “last-call-return” is the last call return feature
“acd-login-logout” “acd-login-logout” is the ACD login/logout feature
acd-agent-available” “acd-agent-available” is the ACD agent

available/unavailable feature
“nway-conference” “nway-conference” is the conference managing feature

Note: For feature.16.name =“nway-conference”:
• If set to 0, the n-way conferencing feature is disabled,
meaning that three-way conferencing can exist, but
there is no manage conference page.
• If set to 1, the n-way conferencing feature is enabled, the
maximum number of conference parties for the platform
can exist, and there is a manage conference page.
Note: The manage conference feature is always disabled on
the SoundPoint IP 32x/33x and 430 phone. The manage
conference feature is always enabled on the SoundStation
IP 7000 and the Polycom VVX 1500 phone.
“call-recording” “call-recording” is the call recording and playback feature
“enhanced-feature-keys” “enhanced-feature-keys” is the enhanced feature keys

feature
Note: This feature must be enabled to use the configurable
soft keys feature.
“corporate-directory” “corporate-directory” is the corporate directory feature
“picture-frame” “picture-frame“ is the digital picture frame feature

Note: feature.20.name = “picture-frame” is only
supported on the Polycom VVX 1500.
feature.x.enabled 0 or 1 (default) except for If set to 0, the feature will be disabled.

x=9 If set to 1, the feature will be enabled and usable by the local
user.
Note feature.16.name =“nway-conference”, feature.17.name =

“call-recording”, and feature.19.name =“corporate-directory” are
charged for separately. To activate these features, you must go to the Polycom
Resource Center (http://extranet.polycom.com/csnprod/signon.html) to retrieve the
activation code. However, these feature are included on the Polycom VVX 1500.
A - 117
Resource <res/>
This attribute’s settings control the maximum size or an external resource
retrieved at run time.
For more information, refer to “Technical Bulletin 35704: Allocating Adequate
Memory for resources on SoundPoint IP and SoundStation IP Phones“ at
• Finder <finder/>
• Quotas <quotas/>
Finder <finder/>
Permitted
res.finder.sizeLimit positive 300 If a resource that is being downloaded to the phone

integer is larger than this value * 1024 bytes (= the
maximum size), the resource will be automatically
truncated to the maximum size defined.
Note: For the SoundPoint IP 550, 560, 650, and 670
phones, this value is internally replaced by 2X the
value. For the SoundStation IP 6000 and 7000
value.
res.finder.minfree 1 to 2048 600 A resource will not be downloaded to the phone if the
amount of free memory is less than this value * 1024
bytes (= the minimum size). This parameter is used
for 16MB SDRAM platforms and scaled up for
platforms with more SDRAM.
If set to 0 or Null, the default value of 600 is used.
Note: For the SoundPoint IP 550, 560, 650, and 670
value. For the SoundStation IP 6000 and 7000
value.
A - 118
Configuration Files
Quotas <quotas/>
Permitted
res.quotas.x.name 1=“tone”, The name of the sub-application for which the particular quota
2= “bitmap”, will apply:
3=“font”, “tone” relates to all downloaded tones and sound effects
5=”background”
“bitmap” relates to all downloaded bitmaps
“font” relates to all downloaded fonts
“background” relates to all downloaded backgrounds
res.quotas.x.value positive integer When a particular resource (one of category “font”, “bitmap”, or
“font”) is downloaded to the phone, a quota equal to this value
* 1024 bytes of compound data size is applied for that
category. If downloading a resource would exceed the quota
for that category, the resource will not be downloaded and a
predefined default will be used instead.
For res.quotas.x.value, the default is 300 KB for tones,
10 KB for bitmaps and fonts, and 600KB for backgrounds.
Note: For the SoundPoint IP 550, 560, 650, and 670 phones,
this value is internally replaced by 2X the value. For the
SoundStation IP 6000 and 7000 phones, this value is internally
replaced by 4X the value.
Microbrowser <mb/>
This attribute’s settings control the home page, proxy and size limits to be used
by the Microbrowser and Browser when it is selected to provide services. The
Microbrowser is supported on the SoundPoint IP 430, 450, 550, 560, 601, 650,
and 670, and the SoundStation IP 6000 and 7000 phones, and the Browser is
supported on the Polycom VVX 1500 phones.
mb.proxy Null or Null. Address of the desired HTTP proxy to be used

domain name or Default by the Microbrowser. If blank, normal unproxied
IP address in the port = HTTP is used by the Microbrowser.
format 8080
<address>:<port>
A - 119
mb.ssawc.enabled 0 or 1 Null If set to 0 or Null, spontaneous display of web

content is disabled.
If set to 1, spontaneous display of web content is
enabled.
mb.ssawc.call.mode Active, Passive Null Control the spontaneous display of web content.
If set to passive or Null, the web content is
displayed only when requested by the user.
If set to active, the web content is displayed
immediately.
• Idle Display <idleDisplay/>
• Main Browser <main/>
• Browser Limits <limits/>
Idle Display <idleDisplay/>

The Microbrowser can be used to create a display that will be part of the
phone’s idle display. These settings control the home page and the refresh rate.
mb.idleDisplay.home Null or any fully Null URL used for Microbrowser idle display home
formed valid HTTP page. For example:
URL. Length up to http://www.example.com/xhtml/frontpage.cgi?pa
255 characters. ge=home. If empty, there will be no
Microbrowser idle display feature. Note that the
Microbrowser idle display will displace the idle
display indicator (refer to
ind.idleDisplay.enabled in Indicators <ind/>
on page A-100).
Note: If ind.idleDisplay.enabled is enabled,
miscellaneous XML errors can occur on
SoundPoint IP 430, 550, 560, 650, and 670 and
SoundStation IP 6000 and 7000 phones.
mb.idleDisplay.refresh 0 or an integer > 5 0 The period in seconds between refreshes of the

idle display Microbrowser's content. If set to 0,
the idle display Microbrowser is not refreshed.
The minimum refresh period is 5 seconds
(values from 1 to 4 are ignored, and 5 is used).
Note: If an HTTP Refresh header is detected, it
will be respected, even if this parameter is set to
0. The refresh parameter will be respected only
in the event that a refresh fails. Once a refresh is
successful, the value in the HTTP refresh
header, if available, will be used.
A - 120
Configuration Files
Main Browser <main/>

This setting controls the home page used by the Microbrowser when that
function is selected.
mb.main.home Any fully formed valid Null URL used for Microbrowser home page. If blank,
HTTP URL. Length the browser will notify the user that a blank
up to 255 characters. home-page was used.
For example:
http://www.example.com/xhtml/frontpage.cgi?pa
ge=home.
mb.main.statusbar 0 or 1 Null Flag to determine whether or not to turn off

display of status messages.
If set to 1, the display of the status bar is
enabled.
If set to 0, or Null, the display of the status bar is
disabled.
mb.main.idleTimeout 0 - 600, seconds Null Timeout for the interactive browser. If the
interactive browser remains idle for a defined
period of time, the phone should return to the
idle browser.
If set to 0, there is no timeout.
If set to Null, the value from up.idleTimeout is
used. Refer to User Preferences <up/> on page
A-31. If mb.main.idleTimeout and
up.idleTimeout are Null, the timeout is 40
seconds.
If set to value greater than 0 and less than 600,
the timeout is for that number of seconds.
mb.main.autoBackKey 0 or 1 1 If set to 1, the phone will automatically supply a

Back soft key in all main browser screens,
which if pressed will take the user back through
the browser history. This is the null default
behavior (for backward compatibility).
If set to 0, the phone will not provide a Back soft
key. All soft keys will be created and controlled
by the application.
A - 121
Browser Limits <limits/>

These settings limit the size of object which the Microbrowser will display by
limiting the amount of memory available for the Microbrowser.
mb.limits.nodes positive integer 256 Limits the number of tags that the XML parser
will handle. This limits the amount of memory
used by complicated pages. A maximum total of
500 (256 each) is recommended. This value is
used as referent values for 16MB of SDRAM.
Note: Increasing this value may have a
detrimental effect on performance of the phone.
mb.limits.cache positive integer 200 Limits the total size of objects downloaded for
each page (both XHTML and images). Once this
limit is reached, no more images are
downloaded until the next page is requested.
Units = kBytes. This value is used as referent
values for 16MB of SDRAM.
Note: Increasing this value may have a
detrimental effect on performance of the phone.
Applications <apps/>
This attribute’s settings control the telephone notification events, state polling
events, and the push server controls. For more information, refer to the Web
Application Developer’s Guide, which can be found at
http://www.polycom.com/voicedocumentation/.
• Telephone Notification <telNotification/>
• State Polling <statePolling/>
• Push <push/>
A - 122
Configuration Files
Telephone Notification <telNotification/>

apps.telNotification.URL URL Null The URL to which the phone sends notifications
of specified events. The protocol used can be
either HTTP or HTTPS.
apps.telNotification. 0 or 1 0 If set to 0, incoming call notification is disabled.

incomingEvent If set to 1, incoming call notification is enabled.
apps.telNotification. 0 or 1 0 If set to 0, outgoing call notification is disabled.

outgoingEvent If set to 1, outgoing call notification is enabled.
apps.telNotification. 0 or 1 0 If set to 0, offhook notification is disabled.

offhookEvent If set to 1, offhook notification is enabled.
apps.telNotification. 0 or 1 0 If set to 0, onhook notification is disabled.

onhookEvent If set to 1, onhook notification is enabled.
State Polling <statePolling/>

apps.statePolling.URL URL Null The URL to which the phone sends call
processing state/device/network information.
The protocol used can be either HTTP or
HTTPS.
Note: To enable state polling, the attributes
apps.statePolling.URL,
apps.statePolling.username, and
apps.statePolling.password must be set to
non-Null values.
apps.statePolling. string Null The user name to access the state polling URL..
username
apps.statePolling. string Null The password to access the state polling URL.
password
A - 123
Push <push/>
apps.push.messageType 0 to 3 0 Select the allowable push priority messages on

phone. The values are:
• 0: (None) Discard push messages
• 1: (Critical) Allows only critical push
messages
• 2: (Normal) Allows only normal push
messages
• 3: (Both) Allows both critical and normal
push messages
apps.push. URL Null The relative URL (received from HTTP URL
serverRootURL Push message) is appended to the application
server root URL and the resultant URL is sent to
the Microbrowser.
For example, if the application server root URL
is http://172.24.128.85:8080/sampleapps and
the relative URL is /examples/sample.html, the
URL that is sent to the Microbrowser is
http://172.24.128.85:8080/sampleapps/example
s/sample.html.
The protocol used can be either HTTP or
HTTPS.
apps.push.username string Null The user name to access the push server URL.
Note: To enable the push functionality, the
attributes apps.push.username and
apps.push.password must be set to non-Null
values.
apps.push.password string Null The password to access the push server URL.
A - 124
Configuration Files
Peer Networking <pnet/>

Peer networking manages communications between Polycom devices. For the
SoundStation IP 7000 conference phone, it manages daisy-chaining and video
integation with the Polycom HDX video systems.
pnet.role “auto” Null The role of the SoundStation IP 7000 when

“standAlone” communicating with other Polycom devices.
“masterOnly” If the attribute is not defined or is null, the default
value is “auto” meaning that the configuration of
“masterPreferred”
the peer role is automatic.
“slaveOnly”
The other value definitions are:
“slavePreferred”
• “standAlone” - IP 7000 is always only
standalone.
• “masterOnly” - IP 7000 is always the master.
• “masterPreferred” - The configuration is
automatic, but if the call capability of the
daisy-chained IP 7000 is the same as this
one, this one is the master.
• “slaveOnly” - IP 7000 is always the slave.
• “slavePreferred” - The configuration is
automatic, but if the call capability of the
daisy-chained IP 7000 is the same as this
one, this one is the slave.
pnet.hdx.ext string Null The HDX Extension Number to be displayed on

the IP 7000 when it is connected to an HDX
system.
pnet.remoteCall. -60 to 0 Null The attenuation applied to tones played by the

callProgAtten IP 7000 for POTS calls when it is connected to
an HDX system when the HDX is the active
speaker.
If set to Null, the default is -15.
pnet.remoteCall. 0 or 1 Null A flag to determine whether or not a dialtone is

localDialTone played when the IP 7000 makes an outgoing
POTS call when it is connected to an HDX.
If set to 1, a dial tone is played.
If set to 0 or Null, a dial tone is not played.
A - 125
DNS Cache <dns/>

In the tables below, a maximum of 12 entries of NAPTR, SRV, and A record
can be added.
• NAPTR <NAPTR/> attribute
• SRV <SRV/>
• A <A/>
NAPTR <NAPTR/>
dns.cache.NAPTR.x. domain name string Null The domain name to which this resource record
name refers.
dns.cache.NAPTR.x.ttl 300 to 65535 300 Specifies the time interval (in seconds) that the
resource record may be cached before the
source of the information should again be
consulted.
dns.cache.NAPTR.x. 0 to 65535 0 A 16-bit unsigned integer specifying the order in

order which the NAPTR records must be processed to
ensure the correct ordering of rules.
dns.cache.NAPTR.x. 0 to 65535 0 A 16-bit unsigned integer that specifies the order

preference in which NAPTR records with equal "order"
values should be processed, low numbers being
processed before high numbers.
dns.cache.NAPTR.x. string Flags to control aspects of the rewriting and

flags interpretation of the fields in the record. Flags
are single characters from the set [A-Z, 0-9].
The alphabetic characters are case insensitive.
At this time only four flag, "S", “A”, “U”, and “P”
are defined. For more information, go to
dns.cache.NAPTR.x. string Specifies the service(s) available down this

service rewrite path. For more information, go to
A - 126
Configuration Files
dns.cache.NAPTR.x. string Null A string containing a substitution expression that

regexp is applied to the original string held by the client
in order to construct the next domain name to
lookup. The grammar of the substitution
expression is given in RFC 2915.
Note: This parameter is currently unused.
dns.cache.NAPTR.x. domain name string Null The next name to query for NAPTR, SRV, or
replacement with SRV prefix address records depending on the value of the
flags field. It must be a fully qualified
domain-name.
SRV <SRV/>
dns.cache.SRV.x.name domain name string Null The domain name string with SRV prefix.
dns.cache.SRV.x.ttl 0 to 65535, seconds 300 Specifies the time interval that the resource
record may be cached before the source of the
information should again be consulted.
dns.cache.SRV.x.priority 0 to 65535 0 The priority of this target host. For more

information, go to
dns.cache.SRV.x.weight 0 to 65535 0 A server selection mechanism. For more

information, go to
dns.cache.SRV.x.port 0 to 65535 0 The port on this target host of this service. For
more information, go to
dns.cache.SRV.x.target domain name string Null The domain name of the target host. For more
information, go to
A - 127
A <A/>
dns.cache.A.x.name valid hostname Null Hostname
dns.cache.A.x.ttl 0 to 65535 300 Specifies the time interval that the resource
record may be cached before the source of the
information should again be consulted.
dns.cache.A.x.address dotted-decimal IP Null IP address that hostname dns.cache.A.x.name

version 4 address maps to.
Soft Keys <softkey/>
Note feature.20.name = “enhanced-feature-keys” must be enabled to use the

Configurable Soft Key feature. Refer to Feature <feature/> on page A-116.
This configuration attribute is defined as follows (where x =1 to maximum

number of defined soft keys):
Permitted
softkey.x.label string Null This is the text displayed with the soft key.
If set to Null, the label to display is
determined as follows:
• If the soft key is mapped to a enhanced
feature key macro, the label of the
enhanced feature key macro will be
used.
• If the soft key is mapped to a speed
dial, the label of the corresponding
directory entry will be used. If this label
does not exist as well and the directory
entry is a enhanced feature key macro,
then the label of the enhanced feature
key macro will be used.
• If the soft key is mapped to chained
actions, only the first one is considered
for label, using the rules above.
• If no labels are found after the above
steps, the soft key label will be blank.
softkey.x.action string Null The same syntax as the enhanced feature

key action. For more information, refer to
Macro Definition on page 4-42.
A - 128
Configuration Files
Permitted
softkey.x.enable 0 (default) Null If set to 0 or Null, the soft key is disabled.

1 If set to 1, the soft key is enabled.
softkey.x.precede 0 (default) Null If set to 0 or Null, the soft key replaces any
1 empty space from the leftmost position.
If set to 1, the soft key is displayed before
the first standard soft key.
softkey.x.use.idle 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the idle state.
If set to 1, the soft key is displayed in the
idle state.
softkey.x.use.active 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the active call state.
active call state.
softkey.x.use.alerting 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the alerting state.
alerting state.
softkey.x.use.dialtone 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the dialtone state.
dialtone state.
softkey.x.use.proceeding 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the proceeding state.
proceeding state.
softkey.x.use.setup 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the setup state.
setup state.
softkey.x.use.hold 0 (default) Null If set to 0 or Null, the soft key is not

1 displayed in the hold state.
hold state.
softkey.feature.newcall 0 Null If set to 0, the New Call soft key is not

1 (default) displayed when there is another way to
place a call.
If set to 1 or Null, the New Call soft key is
displayed.
A - 129
Permitted
softkey.feature.endcall 0 Null If set to 0, the End Call soft key is not

1 (default) displayed.
If set to 1 or Null, the EndCall soft key is
displayed.
softkey.feature.split 0 Null If set to 0, the Split soft key is not

If set to 1 or Null, the Split soft key is
displayed.
softkey.feature.join 0 Null If set to 0, the Join soft key is not

If set to 1 or Null, the Join soft key is
displayed.
softkey.feature.forward 0 Null If set to 0, the Forward soft key is not

If set to 1 or Null, the Forward soft key is
displayed.
softkey.feature.directories 0 Null If set to Null, the Dir soft key is displayed

1 on the SoundPoint IP 32x/33x phone, but
Null (default) not on any other phone.
If set to 0, the Dir soft key is not displayed
on any phone.
If set to 1, the Dir soft key is displayed on
all phones as follows:
• In the idle state, it is displayed after the
New Call and Callers soft keys.
• In the dialtone state, it is displayed after
the End Call and Callers soft keys.
• During a conference or transfer, it is
displayed after the Callers and Cancel
soft keys.
A - 130
Configuration Files
Permitted
softkey.feature.callers 0 Null If set to Null, the Callers soft key is

1 displayed on the SoundPoint IP 32x/33x
Null (default) phone, but not on any other phone.
If set to 0, the Callers soft key is not
displayed on any phone.
If set to 1, the Callers soft key is displayed
on all phones as follows:
• In the idle state, it is displayed after the
New Call soft key and before the Dir
soft key.
• In the dialtone state, it is displayed after
the End Call soft key and before the
Dir soft key.
• During a conference or transfer, it is
displayed before the Cancel soft key.
softkey.feature.mystatus 0 or 1 1 If set to 0, the MyStatus soft key is not

displayed.
If set to 1 or Null, the MyStatus soft key is
displayed.
Note: pres.idleSoftKeys must be set to
1 for this soft key to be displayed.
softkey.feature.buddies 0 or 1 1 If set to 0, the Buddies soft key is not

displayed.
If set to 1 or Null, the Buddies soft key is
displayed.
Note: pres.idleSoftKeys must be set to
1 for this soft key to be displayed.
softkey.feature. 0 or 1 1 If set to 0 and the phone has hard keys

basicCallManagement.redundant mapped for Hold, Transfer, and
Conference functions (all must be
mapped), all of these soft keys are not
displayed.
If set to 1 or Null, all of these soft keys are
displayed.
A - 131
LCD Power Saving <powerSaving/>
Note This attribute is supported on the Polycom VVX 1500 only.
Permitted
powerSaving.enabled 0 or 1 1 If set to 1 or Null, the LCD power saving

feature is enabled.
If set to 0, the LCD power saving feature is
disabled.
powerSaving.officeHours. 0 to 23 8 The starting hour for the day’s office hours,

startHour.xxx where xxx is one of “monday”, “tuesday”,
“wednesday”, “thursday”, “friday”, “saturday”,
and “sunday”.
powerSaving.officeHours. 0 to 12 10 or 0 The duration of the day’s office hours, where

duration.xxx xxx is one of “monday”, “tuesday”,
“wednesday”, “thursday”, “friday”, “saturday”,
and “sunday” .
If set to Null, the default value for the week
days is 10 (hours) and the default value for
Saturday and Sunday is 0 (hours).
powerSaving.idleTimeout. 1 to 600 10 The office hours mode idle timeout (in

officeHours minutes).
powerSaving.idleTimeout. 1 to 10 1 The off hours mode idle timeout (in minutes).

offHours If set to Null, the default value is 1.
powerSaving.idleTimeout. 1 to 20 10 The minimum idle timeout after user input

userInputExtension events (in minutes).
A - 132
Configuration Files
Permitted
powerSaving. 0 to 10 7 The sensitivity of the algorithm used to detect

userDetectionSensitivity. the presence of the phone's user during office
officeHours hours.
If set to 0, this feature is disabled.
If set to Null, the default value is 7. This value
was chosen for good performance in a
typically office environment and is biased for
easy detection during office hours.
powerSaving. 0 to 10 2 The sensitivity of the algorithm used to detect

userDetectionSensitivity. the presence of the phone's user during off
offHours hours.
If set to 0, this feature is disabled.
If set to Null, the default value is 2. This value
was chosen for good performance in a
typically office environment and is biased for
difficult detection during off hours.
Per-Phone Configuration
This section covers the parameters in the per-phone example configuration file
phone1.cfg. This file would normally be used as a template for the per-phone
configuration files. For more information, refer to Deploying Phones From the
Provisioning Server on page 3-17.
The parameters include:
• Registration <reg/>
• Calls <call/>
• Diversion <divert/>
• Dial Plan <dialplan/>
• Messaging <msg/>
• Network Address Translation <nat/>
• Attendant <attendant/>
A - 133
• Roaming Buddies <roaming_buddies/>
• Roaming Privacy <roaming_privacy/>
• User Preferences <up/>
• Automatic Call Distribution <acd/>
Registration <reg/>
SoundPoint IP 32x/33x and 430 support a maximum of two unique
registrations, SoundPoint IP 450 supports three, the SoundPoint IP 550 and 560
supports four, and SoundPoint IP 650 and 670 and the Polycom VVX 1500
support six. Up to three SoundPoint IP Expansion Modules can be added to a
single host SoundPoint IP 650 and 670 phone increasing the total number of
buttons to 34 registrations on the IP 650 and 670. Each registration can
optionally be associated with a private array of servers for completely
segregated signaling. The SoundStation IP 6000, and 7000 supports a single
registration.
In the following table, x is the registration number. IP 32x/33x, 430: x=1-2; IP
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
Permitted
reg.x.csta 0 or 1 Null If set to 1, uaCSTA is enabled.

If reg.x.csta is not Null, this attribute
overrides the global CSTA flag in the sip.cfg
configuration file.
reg.x.displayName UTF-8 encoded Null Display name used in SIP signaling as the
string default caller ID.
Display name used in SIP signaling and/or
H.323 alias as the default caller ID.
reg.x.address string in the format Null The user part or the user and the host part of
userPart from the phone’s SIP URI or the H.323
userPart@domain ID/extension.
For example (SIP): reg.x.address=”1002”
from 1002@polycom.com or
reg.x.address=”1002@polycom.com”.
For example (H.323): reg.x.address=”23456”
reg.x.label UTF-8 encoded Null Text label to appear on the display adjacent
string to the associated line key. If omitted, the label
will be derived from the user part of
reg.x.address.
reg.x.lcs 0 or 1 0 If set to 1, the Microsoft Live Communications

Server is supported for registration x.
A - 134
Configuration Files
Permitted
reg.x.type private OR shared private If set to private, use standard call signaling.
If set to shared, augment call signaling with
call state subscriptions and notifications and
use access control for outgoing calls.
reg.x.thirdPartyName string in the same Null This field must match the reg.x.address
format as value of the other registration which makes
reg.x.address up the bridged line appearance (BLA). It must
be Null in all other cases.
reg.x.auth.userId string Null User ID to be used for authentication

challenges for this registration. If non-Null,
will override the “Reg User x” parameter
entered into the Authentication submenu off
of the Settings menu on the phone.
reg.x.auth.password string Null Password to be used for authentication

challenges for this registration. If non-Null,
will override the “Reg Password x” parameter
entered into the Authentication submenu off
of the Settings menu on the phone.
reg.x.acd-login-logout 0 or 1 0 If both parameters are set to 1 for a

registration, the ACD feature will be enabled
reg.x.acd-agent-available 0 or 1 0 for that registration.
reg.x.ringType 1 to 22 2 The ringer to be used for calls received by

this registration. Default is the first non-silent
ringer.
reg.x.lineKeys 1 to max 1 max = the number of line keys on the phone.

max = 1 on IP 6000, 7000,
max = 2 on IP 32x/33x, 430,
max = 3 on IP 450,
max = 4 on IP 550, 560,
max = 6 on VVX 1500,
max = 34 on IP 650, 670 (without any
Expansion Modules attached, only 6 line keys
are available)
The number of line keys on the phone to be
associated with registration ‘x’.
A - 135
Permitted
reg.x.callsPerLineKey 1 to 34 OR 34 OR For the SoundPoint IP 650 and 670, the

1 to 24 OR 24 OR permitted range is 1 to 34 and the default is
8 OR 4 34.
1 to 8 OR
For the SoundPoint IP 550 and 560 and the
1 to 4
VVX 1500, the permitted range is 1 to 24 and
the default is 24.
For the SoundPoint IP 430 the permitted
range is 1 to 4 and the default is 4.
For all other phones the permitted range is 1
to 8 and the default is 8.
This is the number of calls or conferences
which may be active or on hold per line key
associated with this registration.
Note that this overrides
call.callsPerLineKey for this registration.
Refer to Call Handling Configuration <call/>
on page A-80.
If reg.1.callsPerLineKey is set to 1, call
waiting can be disabled.
Note: A call active on another phone on a
shared line counts as a call for every phone
sharing that registration.
reg.x.bargeInEnabled 0 or 1 Null Allow remote user of SCA to interrupt call.

(Works in a similar way to resume.)
If set to 1, barge-in is enabled for line x.
If set to 0 or Null, barge-in is disabled for line
x.
reg.x.outboundProxy.address dotted-decimal IP Null IP address or host name and port of a SIP

address or host server to which the phone shall send all
name requests.
reg.x.outboundProxy.port 1 to 65535 5060
A - 136
Configuration Files
Permitted
reg.x.outboundProxy.transport DNSnaptr or DNSnap If set to Null or DNSnaptr:

TCPpreferred or tr If reg.x.outboundProxy.address is a
UDPOnly or hostname and reg.x.outboundProxy.port is 0
TLS or or Null, do NAPTR then SRV look-ups to try
TCPOnly to discover the transport, ports and servers,
as per RFC 3263. If
reg.x.outboundProxy.address is an IP
address, or a port is given, then UDP is used.
TCP fails.
If set to UDPOnly:
If set to TLS:
If set to TCPOnly:
reg.x.proxyRequire string Null The string that needs to appear in the

“Proxy-Require” header. If Null, no
"Proxy-Require" will be sent.
reg.x.serverFeatureControl.cf 0 or 1 0 If set to 1, server-based call forwarding is

enabled. The call server has control of call
forwarding.
If set to 0, server-based call forwarding is not
enabled. This is the old behavior.
If reg.x.serverFeatureControl.cf is not
Null, this attribute overrides the global
server-based call forwarding flag in the
sip.cfg configuration file.
reg.x.serverFeatureControl.dnd 0 or 1 0 If set to 1, server-based DND is enabled. The

call server has control of DND.
If set to 0, server-based DND is not enabled.
This is the old behavior.
If reg.x.serverFeatureControl.dnd is not
Null, this attribute overrides the global
server-based call forwarding flag in the
A - 137
Permitted
reg.x.auth.optimizedInFailover 0 or 1 0 If set to 1, when failover occurs, the first new

SIP request is sent to the server that sent the
proxy authentication request.
If set to 0, when failover occurs, the first new
SIP request is sent to the server with the
highest priority in the server list.
If this parameter is Null,
voIpProt.SIP.authOptimizedInFailover
is checked.
If both parameters are set, this parameter
takes precedence.
reg.x.strictLineSeize 0 or 1 Null If set to 1, forces phone to wait for 200 OK on

registration x when receiving a TRYING
notify.
voIpProt.SIP.strictLineSeize is
checked.
takes precedence.
reg.x.musicOnHold.uri string Null A URI that provides the media stream to play
for the remote party on hold.
When present, and if reg.x.musicOnHold is
not Null, this attribute overrides the global
Music on Hold defined in the sip.cfg
configuration file.
reg.x.tcpFastFailover 0 or 1 Null If set to 1, failover occurs based on the

values of reg.x.server.y.retryMaxCount
voIpProt.server.x.retryTimeOut.
voIpProt.SIP.tcpFastFailover is
checked.
takes precedence.
reg.x.protocol.SIP 0 or 1 1 If set to 1, SIP signaling is enabled for this

line registration.
If set to 0, SIP signaling is not enabled for this
line registration.
reg.x.protocol.H323 0 or 1 0 If set to 1, H.323 signaling is enabled for this

line registration.
If set to 0, H.323 signaling is not enabled for
this line registration.
A - 138
Configuration Files
Permitted
reg.x.server.y.address dotted-decimal IP Null Optional IP address or host name, port,

address or host transport, registration period, fail-over
name parameters and line seize subscription period
of a SIP server that accepts registrations.
reg.x.server.y.port 0, Null, 1 to 65535 Null Multiple servers can be listed starting with
y=1, 2, ... for fault tolerance. If specified,
reg.x.server.y.transport DNSnaptr or DNSnap
these servers may override the servers
TCPpreferred or tr
specified in sip.cfg in Server <server/> on
UDPOnly or
page A-8.
TLS or
TCPOnly Note: If the reg.x.server.y.address parameter
is non-Null, all of the reg.x.server.y.xxx
reg.x.server.y.expires positive integer Null parameters will override the parameters
specified in sip.cfg in Server <server/> on
reg.x.server.y.register 0 or 1 Null page A-8.
reg.x.server.y.expires.overlap positive integer, 60 Note: If the reg.x.server.y.address parameter
minimum 5, is non-Null, it takes precedence even if the
maximum 65535 DHCP server is available.
reg.x.server.y.retryTimeOut Null or Null

non-negative
integer
reg.x.server.y.retryMaxCount Null or Null

non-negative
integer
reg.x.server.y.expires.lineSeize positive integer Null
reg.x.server.y.lcs 0 or 1 0 This attribute overrides the reg.x.lcs.

If set to 1, the Microsoft Live Communications
Server is supported for registration x.
reg.x.server.H323.y.address dotted-decimal IP Null Address of the H.323 gatekeeper.

address or host
name
reg.x.server.H323.y.port 0, Null, 1 to 65535 Null Port to be used for H.323 signaling.

If set to Null, 1719 (H.323 RAS signaling) is
used.
reg.x.server.H323.y.expires postive integer Null Desired registration period.
Calls <call/>
This attribute affects the call-oriented per-phone configuration.
• Do Not Disturb <donotdisturb/>
• Automatic Off-Hook Call Placement <autoOffHook/>
A - 139
• Missed Call Configuration <serverMissedCall/>
• Missed Call Tracking <missedCallTracking/>
• Call Waiting <callWaiting/>
Do Not Disturb <donotdisturb/>

Permitted
call.donotdisturb.perReg 0 or 1 0 If set to 1, the DND feature will allow selection of

DND on a per-registration basis.
NOTE: If
voIpProt.SIP.serverFeatureControl.dnd is
set to 1 (enabled), this parameter is ignored. For
more information, refer to SIP <SIP/> on page
A-11.
Automatic Off-Hook Call Placement <autoOffHook/>

An optional per-registration feature is supported which allows automatic call
placement when the phone goes off-hook.
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6;
IP 650, 670: x=1-34; IP 6000: x=1; IP 7000: x=1.
call.autoOffHook.x.enabled 0 or 1 0 If set to 1, a call will be automatically

placed to the contact specified upon
call.autoOffHook.x.contact ASCII encoded string Null going off-hook on this registration.
containing digits (the user part
of a SIP URL) or a string that
constitutes a valid SIP URL
(6416 or 6416@polycom.com)
call.autoOffHook.x.protocol “SIP” or “H323” Null On a dual-protocol line only,

specifies the routing protocol to use
for the auto off-hook dialing. The
strings are case sensitive.
If set to Null, the value of
call.autoRouting.preferredProt
ocol is used.
Note: If a line is single-protocol
configured, the configured protocol
will be used in the auto off-hook
dialing and any value in its
call.autoOffHook.x.protocol
field will be ignored.
A - 140
Configuration Files
Missed Call Configuration <serverMissedCall/>

The phone supports a per-registration configuration of which events will
cause the locally displayed “missed calls” counter to be incremented.
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
Permitted
call.serverMissedCall.x.enabled 0 or 1 0 If set to 0, all missed-call events will increment

the counter.
If set to 1, only missed-call events sent by the
server will increment the counter.
NOTE: This feature is supported with the
Sylantro call server only.
Missed Call Tracking <missedCallTracking/>

You can enable/disable missed call tracking on a per-line basis.
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
Permitted
call.missedCallTracking.x.enabled 0 or 1 1 If set to 1 or Null, missed call tracking is

enabled.
If call.missedCallTracking.x.enabled is
set to 0, then missedCall counter is not
updated regardless of what
call.serverMissedCalls.x.enabled is set
to (and regardless of how the server is
configured). There is no Missed Call List
provided under Menu > Features of the phone.
set to 1 and call.serverMissedCalls.x.enabled
is set to 0, then the number of missedCall
counter is incremented regardless of how the
server is configured.
set to 1 and
call.serverMissedCalls.x.enabled is set
to 1, then the handling of missedCalls depends
on how the server is configured.
A - 141
Call Waiting <callWaiting/>

Permitted
call.callWaiting.ring beep, ring, beep Specifies the ring tone heard on an incoming
silent call when another call is active.
If set to Null, the default value is beep.
Diversion <divert/>
The phone has a flexible call forward/diversion feature for each registration.
In all cases, a call will only be diverted if a non-Null contact has been
configured.
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
divert.x.contact ASCII encoded string Null The forward-to contact used for
containing digits (the user all automatic call diversion
part of a SIP URL) or a string features unless overridden by a
that constitutes a valid SIP specific contact of a per-call
URL (6416 or diversion feature (refer to
6416@polycom.com below).
divert.x.autoOnSpecificCaller 0 or 1 1 If set to 1, calls may be diverted

using the Auto Divert feature of
the directory. This is a global
flag.
Note: If server-based call
forwarding is enabled, this
divert.x.sharedDisabled 0 or 1 1 If set to 1, all diversion features

on that line will be disabled if
the line is configured as
shared.
• Forward All <fwd/>
• Busy <busy/>
• No Answer <noanswer/>
• Do Not Disturb <dnd/>
A - 142
Configuration Files
Forward All <fwd/>

Permitted
divert.fwd.x.enabled 0 or 1 1 If set to 1, the user will be able to enable universal call

forwarding through the soft key menu.
Note: If server-based call forwarding is enabled, this
parameter is enabled.
Busy <busy/>
Calls can be automatically diverted when the phone is busy.
divert.busy.x.enabled 0 or 1 1 If set to 1, calls will be

forwarded on busy to the
contact specified below.
divert.busy.x.timeout positive integer 60 Time in seconds to allow

altering before initiating the
diversion.
divert.busy.x.contact ASCII encoded string Null Forward-to contact for calls

containing digits (the user part forwarded due to busy status, if
of a SIP URL) or a string that Null, divert.x.contact will be
constitutes a valid SIP URL used.
(6416 or 6416@polycom.com
A - 143
No Answer <noanswer/>
The phone can automatically divert calls after a period of ringing.
divert.noanswer.x.enabled 0 or 1 1 If set to 1, calls will be

forwarded on no answer to the
contact specified.
divert.noanswer.x.timeout positive integer 55 Time in seconds to allow

altering before initiating the
diversion.
divert.noanswer.x.contact ASCII encoded string Null Forward-to contact used for

containing digits (the user part calls forwarded due to no
of a SIP URL) or a string that answer, if Null,
constitutes a valid SIP URL divert.x.contact will be
(6416 or 6416@polycom.com) used.
Do Not Disturb <dnd/>

The phone can automatically divert calls when Do Not Disturb (DND) is
enabled.
divert.dnd.x.enabled 0 or 1 0 If set to 1, calls will be

forwarded on DND to the
contact specified below.
Note: If server-based DND or
server-base call forwarding is
enabled, this parameter is
disabled.
divert.dnd.x.contact ASCII encoded string containing digits Null Forward-to contact used for
(the user part of a SIP URL) or a string calls forwarded due to DND
that constitutes a valid SIP URL (6416 or status, if Null
6416@polycom.com) divert.x.contact will be
used.
A - 144
Configuration Files
Dial Plan <dialplan/>

Per-registration dial plan configuration is supported.
450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
Permitted
dialplan.x.applyToCallListDial 0 or 1 0 When present, and if

dialplan.x.digitmap is not
Null, this attribute overrides the
global dial plan defined in the
For interpretation, refer to Dial
Plan <dialplan/> on page A-23.
dialplan.x.applyToDirectoryDial 0 or 1 0 When present, and if

dialplan.x.applyToUserDial 0 or 1 1 When present, and if

dialplan.x.applyToUserSend 0 or 1 1 When present, and if

dialplan.x.impossibleMatchHandling 0, 1 or 2 0 When present, and if

A - 145
Permitted
dialplan.x.removeEndOfDial 0 or 1 1 When present, and if

dialplan.x.applyToTelUriDial 0 or 1 1 When present, and if

• Digit Map <digitmap/>
• Routing <routing/>
A - 146
Configuration Files
Digit Map <digitmap/>

For more information on digit map syntax, refer to Digit Map <digitmap/> on
page A-24.
Permitted
dialplan.x.digitmap A string compatible with the Null When present, this attribute
digit map feature of MGCP overrides the global dial plan
described in 2.1.5 of RFC defined in the sip.cfg
3435; string is limited to 768 configuration file.
bytes and 30 segments; a For more information, refer to
comma is also allowed; a Digit Map <digitmap/> on page
comma is also allowed; A-24.
when reached in the digit
map, a comma will turn dial
tone back on;’+’ is allowed
as a valid digit; extension
letter ‘R’ is used as defined
above.
dialplan.x.digitmap.timeOut string of positive integers Null When present, and if

separated by ‘|’ dialplan.x.digitmap is not
Digit Map <digitmap/> on page
A-24.
Routing <routing/>
This attribute allows specific routing paths for outgoing SIP calls to be
configured independent of other ‘default’ configuration.
• Emergency <emergency/>
A - 147
Server <server/>
dialplan.x.routing.server.y. dotted-decimal IP address Null IP address or host name and

address or host name port of a SIP server that will
be used for routing calls.
dialplan.x.routing.server.y.port 1 to 65535 5060 Multiple servers can be listed
starting with y=1, 2, ... for
fault tolerance.
Emergency <emergency/>
In the following attributes, y is the index of the emergency entry description
and z is the index of the server associated with the emergency entry y. For each
emergency entry (index y), one or more server entry (indexes (y,z)) can be
configured. y and z must both follow single step increasing numbering starting
at 1.
dialplan.x.routing.emergency. Comma separated list of Null This represents the URLs

y.value entries or single entry Example: that should be watched for
representing a or a “15,17,18”, “911”, emergency routing.
combination of SIP URL. “sos”. When one of these defined
URL is detected as being
dialed by the user, the call
will be automatically directed
to the defined emergency
server.
dialplan.x.routing.emergency. positive integer Null Index representing the

y.server.z server defined in Server
<server/> on page A-148
that will be used for
emergency routing.
A - 148
Configuration Files
Messaging <msg/>
Message-waiting indication is supported on a per-registration basis.
Permitted
msg.bypassInstantMessage 0 or 1 0 If set to 1, the display offering a choice of

“Message Center” and “Instant Messages” will
be bypassed when pressing the Messages key.
The phone will act as if “Message Center” was
chosen. Refer to Voice Mail Integration on page
4-56. Instant Messages will still be accessible
from the Main Menu.
• Message Waiting Indicator <mwi/>
Message Waiting Indicator <mwi/>

450: x=1-3; IP 550, 560: x=1-4; VVX 1500: x=1-6; IP 650, 670: x=1-34; IP 6000: x=1;
IP 7000: x=1.
A - 149
msg.mwi.x.subscribe ASCII encoded string containing Null If non-Null, the phone will send
digits (the user part of a SIP a SUBSCRIBE request to this
URL) or a string that constitutes contact after boot-up.
a valid SIP URL (6416 or
6416@polycom.com)
msg.mwi.x. contact or “registration” Configures message retrieval

callBackMode registration or and notification for the line.
disabled If set to “contact”, a call will be
placed to the contact specified
in the callback attribute when
the user invokes message
retrieval.
If set to “registration”, a call will
be placed using this registration
to the contact registered (the
phone will call itself).
If set to “disabled”, message
retrieval and message
notification are disabled.
msg.mwi.x.callBack ASCII encoded string containing Null Contact to call when retrieving
digits (the user part of a SIP messages for this registration.
URL) or a string that constitutes
a valid SIP URL (6416 or
6416@polycom.com)
Network Address Translation <nat/>

These parameters define port and IP address changes used in NAT traversal.
The port changes will change the port used by the phone, while the IP entry
simply changes the IP advertised in the SIP signaling. This allows the use of
simple NAT devices that can redirect traffic, but do not allow for port
mapping. For example, port 5432 on the NAT device can be sent to port 5432
on an internal device, but not port 1234.
A - 150
Configuration Files
Permitted
nat.ip dotted-decima Null IP address to advertise within SIP signaling - should

l IP address match the external IP address used by the NAT device.
nat.signalPort 1024 to 65535 Null If non-Null, this port will be used by the phone for SIP
signaling, overriding the value set for
voIpProt.local.Port in sip.cfg.
nat.mediaPortStart 1024 to 65535 Null If non-Null, this attribute will be used to set the initially
allocated RTP port, overriding the value set for
tcpIpApp.port.rtp.mediaPortRangeStart in sip.cfg.
Refer to RTP <rtp/> on page A-78.
nat.keepalive.interval 0 to 3600 Null If non-Null (or 0), the keepalive interval in seconds. This
parameter is used to set the interval at which phones will
send a keep-alive packet to the gateway/NAT device to
keep the communication port open so that NAT can
continue to function as setup initially.
The Microsoft Live Communications Server 2005
keepalive feature will override this interval. If you want to
deploy phones behind a NAT and connect them to Live
Communications Server, the keepalive interval received
from the Live Communications Server must be short
enough to keep the NAT port open. Once the TCP
connection is closed, the phones stop sending keep-alive
packets.
Attendant <attendant/>
Note These attributes are available on SoundPoint IP 32x/33x, 430, 450, 550, 560, 650,
and 670 phones only.
The Busy Lamp Field (BLF) / attendant console feature enhances support for
a phone-based attendant console.
A - 151
Permitted
attendant.uri string Null For attendant console / busy lamp field (BLF) feature.
This specifies the list SIP URI on the server. If this is just
a user part, the URI is constructed with the server host
name/IP.
Note: If attendant.uri is set, then the individually
addressed users configured by
attendant.resourceList and attendant.behaviors
attributes are ignored.
attendant.reg positive 1 For attendant console / BLF feature. This is the index of
integer the registration which will be used to send a SUBSCRIBE
to the list SIP URI specified in attendant.uri. For example,
attendant.reg = 2 means the second registration will
be used.
attendant.ringType 1 to 22 1 The ring tone to play when a BLF dialog is in the offering
state.
• Resource List <resourceList/>
• Behaviors <behaviors/>
Resource List <resourceList/>

In the following table, x is the monitored user number. For IP 450: x=1-2; IP
550, IP 560: X=1-3; IP 650, IP 670: x=1-47.
attendant.resourceList.x.address string that Null The user referenced by

constitutes a valid attendant.reg="" will subscribe to this
SIP URI URI for dialog. If a user part is present, the
(sip:6416@polyco phone will subscribe to a sip URI
m.com) or contains constructed from user part and the domain
the user part of a of the user referenced by attendant.reg.
SIP URI (6416)
attendant.resourceList.x.label UTF-8 encoded Null Text label to appear on the display

string adjacent to the associated line key. If set
to Null, the label will be derived from the
user part of
attendant.resourceList.x.address .
A - 152
Configuration Files
attendant.resourceList.x.type "normal" or “normal” Type of resource being monitored.

"automata" If set to normal, the default action when
pressing the line key adjacent to this
monitored user is to initiate a call if the
user is idle or busy and to perform a
directed call pickup if the user is ringing.
Any active calls are first placed on hold.
If set to automata, the default action when
pressing the line key adjacent to this
monitored user is to perform a park/blind
transfer of any currently active call. If there
is no active call and the monitored user is
ringing/busy, an attempt to perform a
directed call pickup/park retrieval is made.
Behaviors <behaviors/>
attendant.behaviors.display. 0 or 1 1 A flag to determine whether or

spontaneousCallAppearances.normal not a call appearance is
spontaneously presented to the
attendant.behaviors.display. 0 or 1 0 attendant when calls are alerting
spontaneousCallAppearances.automata on a monitored resource. The
information displayed after a
press-and-hold of a resource's
line key is unchanged by this
parameter. If set to 1, the display
is enabled.
attendant.behaviors.display. 0 or 1 1 A flag to determine whether or

remoteCallerID.normal not remote party caller ID
information is presented to the
attendant.behaviors.display. 0 or 1 1 attendant. If set to 0 (disabled),
remoteCallerID.automata the string "unknown" would be
substituted for both name and
number information.
A - 153
Roaming Buddies <roaming_buddies/>
Note This attribute is used in conjunction with Microsoft Live Communications

Server 2005 only.
Permitted
roaming_buddies.reg positive Null Specifies the line/registration number which has roaming
integer buddies support enabled. If Null, roaming buddies is
disabled. If value < 1, then value is replaced with 1.
Warning: This parameter must be enabled
(value > 0) if the call server is Microsoft Live
Communications Server 2005.
Roaming Privacy <roaming_privacy/>
Note This attribute is used in conjunction with Microsoft Live Communications

Server 2005 only.
Permitted
roaming_privacy.reg positive Null Specifies the line/registration number which has roaming
integer privacy support enabled. If Null, roaming privacy is
disabled. If value < 1, then value is replaced with 1.
A - 154
Configuration Files
User Preferences <up/>

Permitted
up.analogHeadsetOption 0, 1, or 2 0 Selects optional external hardware for use with a

headset attached to the phone's analog headset jack.
If set to 0, no compatible headset is attached.
If set to 1, a DHSG-compatible headset is attached
and can be used as an electronic hookswitch.
If set to 2, a Plantronics compatible headset is
attached and can be used an electronic hookswitch.
up.offHookAction.none 0 or 1 Null If set to 0 or Null, the behavior introduced in SIP 2.1.2

occurs. When users go off-hook, the phone tries to
seize a line. Which line is seized depends on
voIpProt.SIP.strictLineSeize,
voIPProt.SIP.lineSeize.retries, and
reg.x.strictLineSeize.
If set to 1, the behavior from SIP 1.6.7 occurs. When
users go off-hook, the phone does not seize a line or
answer a ringing call. The user must use the line keys
to either make a new call or answer a ringing call.
This will apply under all ringer settings, not just
SilentRing.
up.pictureFrame.folder string Null The path name for images. The maximum length is 40
characters.
If set to Null, images stored in the root folder on the
USB flash drive are displayed.
For example, if the images are stored in the
“/images/phone” folder on the USB flash drive, set
up.pictureFrame.folder to images/phone .
up.pictureFrame. 3 to 300 Null The time to display the image.

timePerImage seconds If set to Null, the default time is 5 seconds.
A - 155
Permitted
up.screenSaver.enabled 0 or 1 0 If set to 1, a USB flash drive is attached to the phone,

and the idle browser is not configured, a slide show
will cycle through the images from the USB flash
drive. The images must stored in the appropriate
directory of the USB flash drive
(up.pictureFrame.folder in phone1.cfg). The slide
show does not appear when the phone is in the active
state. If set to 1, but there is no USB flash drive
attached to the phone, there is not change on the
screen. However, the screen saver will start working
once a USB flash drive is attached.
If set to 0, the feature is disabled.
Note: If the idle browser is also enabled, the idle
browser is displayed until the screen saver times out;
then the screen saver appears. When the screen
saver exits, the idle browser is displayed again and is
up to date (it is refreshed in the background).
up.screenSaver.waitTime 1 to 9999 Null The time to wait (In minutes) in the idle state (until the
screen saver starts).
If set to Null, the default time is 15 minutes.
Automatic Call Distribution <acd/>

Permitted
acd.reg 1 to 34 1 The registration index used to support BroadSoft

server-based ACD. If set to Null, line 1 is used.
acd.stateAtSignIn 0 or 1 1 The state of the user when signing in.

If set to 1 or Null, the user is available.
If set to 0, the user is unavailable.
A - 156
Configuration Files
Flash Parameter Configuration

Any field in the BootROM setup menu and the application Line Configuration
and Call Server Configuration menus can be set through a configuration file.
A DHCP server can be configured to point the phones to a provisioning server
that has the required configuration files. The new settings will be downloaded
by the phones and used to configure them. This removes the need for manual
interaction with phones to configure basic settings. This is especially useful for
initial installation of multiple phones.
These device settings are detected when the application starts. If the new
settings would normally cause a reboot if they were changed in the application
Network Configuration menu, then they will cause a reboot when the
application starts.
Warning The parameters for this feature should be put in separate configuration files to
simplify maintenance. Do not add them to existing configuration files (such as
sip.cfg). One new configuration file will be required for parameters that should
apply to all phones, and individual configuration files will be required for
phone-specific parameters such as SIP registration information.
The global device.set parameter must be enabled when the initial

installation is done, and then it should be disabled. This prevents subsequent
reboots by individual phones triggering a reset of parameters on the phone
that may have been tweaked since the initial installation.
Warning This feature is very powerful and should be used with caution. For example, an
incorrect setting could set the IP Address of multiple phones to the same value.
Note that some parameters may be ignored, for example if DHCP is enabled it will
still override the value set with device.net.ipAddress.
Individual parameters are checked to see whether they are in range, however, the
interaction between parameters is not checked. If a parameter is out of range, an
error message will appear in the log file and parameter will not be used.
Incorrect configuration could cause phones to get into a reboot loop. For example,
server A has a configuration file that specifies that server B should be used, which
has a configuration file that specifies that server A should be used.
Polycom recommends that you test the new configuration files on two phones
before initializing all phones. This should detect any errors including IP address
conflicts.
A - 157
The flash attributes are defined as follows:
device.set 0 or 1 If set to 0, do not use any device.xxx.yyy fields to

default = 0 set any parameters. Set this to 0 after the initial
installation.
If set to 1, use the device.xxx.yyy fields that have
device.xxx.yyy.set = 1. Set this to 1 for the initial
installation only.
device.xxx.yyy.set 0 or 1 If set to 0, do not use the device.xxx.yyy value.

default = 0 If set to 1, use the device.xxx.yyy value.
For example, if device.net.ipAddress.set = 1,
then use the contents of the device.net.ipAddress
field.
device.net.ipAddress dotted-decimal IP address Phone's IP address.

Note: This field is not used when DHCP client is
enabled.
device.net.subnetMask dotted-decimal IP address Phone's subnet mask.

enabled.
device.net.IPgateway dotted-decimal IP address Phone's default router / IP gateway.

enabled.
device.net.vlanId Null, 0 to 4094 Phone’s 802.1Q VLAN identifier.

Note: Null = no VLAN tagging
device.net.cdpEnabled 0 or 1 If set to 1, the phone will attempt to determine its

VLAN ID and negotiate power through CDP.
device.dhcp.enabled 0 or 1 For description, refer to DHCP or Manual TCP/IP

Setup on page 3-2.
device.dhcp. 0 to 3 For descriptions, refer to DHCP Menu on page 3-8.

bootSrvUseOpt
device.dhcp.bootSrvOpt 128 to 254 (Cannot be the

same as VLAN ID Option)
device.dhcp. 0 or 1
bootSrvOptType
device.dhcp. 0 to 2
dhcpVlanDiscUseOpt
device.dhcp. 128 to 254 (Cannot be the

dhcpVlanDiscOpt same as provisioning server
Option)
A - 158
Configuration Files
device.prov.serverName any string For descriptions, refer to Server Menu on page 3-10.
device.prov.serverType 0 to 4
device.prov.user any string
device.prov.password any string
device.prov.appProvType 0 or 1
device.prov.appProvString any string
device.prov. 10, Null

redunAttemptLimit
device.prov. 300, Null

redunInterAttemptDelay
device.prov. 1 to 8
maxRedunServers
device.sntp.serverName any string Can be dotted-decimal IP address or domain name

string. SNTP server from which the phone will obtain
the current time
device.sntp.gmtOffset -43200 to 46800 GMT offset in seconds, corresponding to -12 to +13

hours.
device.dns.serverAddress dotted-decimal IP address Primary server to which the phone directs Domain
device.dns.altSrvAddress dotted-decimal IP address Secondary server to which the phone directs Domain
device.dns.domain any string The phone’s DNS domain.
device.auth. any string The phone’s local administrator password.

localAdminPassword
device.auth. any string The phone user’s local password.

localUserPassword
device.auth.regUserx any string The SIP registration user name for registration x
where x = 1 to 48.
device.auth.regPasswordx any string The SIP registration password for registration x

where x = 1 to 48.
device.sec. any string Configuration encryption key that is used for

configEncryption.key encryption of configuration files.
device.syslog.serverName dotted-decimal IP address The syslog server IP address or host name.

OR The default value is NULL.
domain name string
A - 159
device.syslog.transport None=0, The protocol that the phone will use to write to the
UDP=1, syslog server.
TCP=2, If set to “None”, transmission is turned off, but the
TLS=3 server address is preserved.
device.syslog.facility 0 to 23 A description of what generated the log message.

For more information, refer to section 4.1.1 of RFC
3165.
The default value is 16, which maps to “local 0”.
device.syslog.renderLevel 0 to 6 Specifies the lowest class of event that will be

rendered to syslog. It is based on
log.render.level and can be a lower value.
Refer to Basic Logging <level/><change/> and
<render/> on page A-106.
device.syslog.prependMac Enabled, Disabled If enabled, the phone’s MAC address is prepended

to the log message sent to the syslog server.
device.em.power Enabled, Disabled, Null Refer to the EM Power parameter in Main Menu on
page 3-7.
device.net.etherVlanFilter Enabled, Disabled Refer to the VLAN Filtering parameter in Ethernet

Menu on page 3-12.
device.net.etherStormFilter Enabled, Disabled Refer to the Storm Filtering parameter in Ethernet

Menu on page 3-12.
device.net.etherModeLAN -1 to 5 Refer to the LAN Port Mode parameter in Ethernet

Menu on page 3-12.
device.net.etherModePC -1 to 5 Refer to the PC Port Mode parameter in Ethernet

Menu on page 3-12.
device.serial.enable 0,1 Enables the debug serial port.

The default value is 1.
device.sec.SSL.certList all, custom, default The type of certificate list.
device.sec.SSL.customCert X.509 certificate The certificate value.
device.net.lldpEnabled 0 or 1 If set to 1, the phone will attempt to determine its

VLAN ID and negotiate power through LLDP.
If set to 0, the phone will not attempt to determine its
VLAN ID or power management through LLDP.
device.prov.clinkEnabled 0 or 1 (default) If set to 1, enable the CLink subsystem when the

Ethernet cable is not attached.
If set to 0, disable the CLink subsystem when the
Ethernet cable is not attached.
A - 160
B
Session Initiation Protocol (SIP)
This chapter provides a description of the basic Session Initiation Protocol

(SIP) and the protocol extensions that are supported by the current SIP
application. To find the applicable Request For Comments (RFC) document,
go to http://www.ietf.org/rfc.html and enter the RFC number.
This chapter contains information on:
• Basic Protocols—All the basic calling functionality described in the SIP

specification is supported. Transfer is included in the basic SIP support.
• Protocol Extensions—Extensions add features to SIP that are applicable to

a range of applications, including reliable 1xx responses and session
timers.
For information on supported RFC’s and Internet drafts, refer to the following
section, RFC and Internet Draft Support.
This chapter also describes:
• Request Support
• Header Support
• Response Support
• Hold Implementation
• Reliability of Provisional Responses
• Transfer
• Third Party Call Control
• SIP for Instant Messaging and Presence Leveraging Extensions
• Shared Call Appearance Signaling
• Bridged Line Appearance Signaling
B-1
RFC and Internet Draft Support

The following RFC’s and Internet drafts are supported:
• RFC 1321—The MD5 Message-Digest Algorithm
• RFC 2327—SDP: Session Description Protocol
• RFC 2387—The MIME Multipart / Related Content-type
• RFC 2976—The SIP INFO Method
• RFC 3261—SIP: Session Initiation Protocol (replacement for RFC 2543)
• RFC 3262—Reliability of Provisional Responses in the Session Initiation

Protocol (SIP)
• RFC 3263—Session Initiation Protocol (SIP): Locating SIP Servers
• RFC 3264—An Offer / Answer Model with the Session Description

Protocol (SDP)
• RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification
• RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method
• RFC 3325—SIP Asserted Identity
• RFC 3515—The Session Initiation Protocol (SIP) Refer Method
• RFC 3555 — MIME Type of RTP Payload Formats
• RFC 3611 — RTP Control Protocol Extended reports (RTCP XR)
• RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples
• draft-ietf-sip-cc-transfer-05.txt—SIP Call Control - Transfer
• RFC 3725—Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)
• RFC 3842—A Message Summary and Message Waiting Indication Event

Package for the Session Initiation Protocol (SIP)
• RFC 3856—A Presence Event Package for Session Initiation Protocol (SIP)
• RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header
• RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism
• RFC 3959—The Early Session Disposition Type for the Session Initiation
Protocol (SIP)
• RFC 3960—Early Media and Ringing Tone Generation in the Session

Initiation Protocol (SIP)
• RFC 3968—The Internet Assigned Number Authority (IANA) Header

Field Parameter Registry for the Session Initiation Protocol (SIP)
B-2
• RFC 3969—The Internet Assigned Number Authority (IANA) Uniform

Resource Identifier (URI) Parameter Registry for the Session Initiation
Protocol (SIP)
• RFC 4028—Session Timers in the Session Initiation Protocol (SIP)
• RFC 4235—An INVITE-Initiated Dialog Event Package for the Session

• draft-levy-sip-diversion-08.txt—Diversion Indication in SIP
• draft-anil-sipping-bla-02.txt—Implementing Bridged Line Appearances

(BLA) Using Session Initiation Protocol (SIP)
• draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller

Identity and Privacy within Trusted Networks
• draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing

for User Agents
• draft-ietf-sipping-rtcp-summary-02.txt —Session Initiation Protocol

Package for Voice Quality Reporting Event
• draft-ietf-sip-connect-reuse-04.txt—Connection Reuse in the Session

Request Support
The following SIP request messages are supported:
Method Supported Notes
REGISTER Yes
INVITE Yes
ACK Yes
CANCEL Yes
BYE Yes
OPTIONS Yes
SUBSCRIBE Yes
NOTIFY Yes
REFER Yes
PRACK Yes
B-3
Method Supported Notes
INFO Yes RFC 2976, the phone does not generate INFO
requests, but will issue a final response upon
receipt. No INFO message bodies are parsed.
MESSAGE Yes Final response is sent upon receipt. Message

bodies of type text/plain are sent and received.
UPDATE Yes
Header Support
The following SIP request headers are supported:
Note In the following table, a “Yes” in the Supported column means the header is sent
and properly parsed.
Header Supported Notes
Accept Yes
Accept-Encoding No
Accept-Language Yes
Access-Network-Info No
Alert-Info Yes
Allow Yes
Allow-Events Yes
Authentication-Info No
Authorization Yes
Call-ID Yes
Call-Info Yes
Contact Yes
Content-Disposition No
Content-Encoding No
Content-Language No
Content-Length Yes
Content-Type Yes
CSeq Yes
B-4
Date No
Diversion Yes
Error-Info No
Event Yes
Expires Yes
From Yes
In-Reply-To No
Max-Forwards Yes
Min-Expires No
Min-SE Yes
MIME-Version No
Organization No
P-Asserted-Identity Yes
P-Preferred-Identity Yes
Priority No
Privacy No
Proxy-Authenticate Yes
Proxy-Authorization Yes
Proxy-Require Yes
RAck Yes
Record-Route Yes
Refer-To Yes
Referred-By Yes
Referred-To Yes
Remote-Party-ID Yes
Replaces Yes
Reply-To No
Requested-By No
Require Yes
Response-Key No
B-5
Retry-After Yes
Route Yes
RSeq Yes
Server Yes
Session-Expires Yes
Subject Yes
Subscription-State Yes
Supported Yes
Timestamp Yes
To Yes
Unsupported Yes
User-Agent Yes
Via Yes
Warning Yes Only warning codes 300 to 399
WWW-Authenticate Yes
Response Support
The following SIP responses are supported:
Note In the following table, a “Yes” in the Supported column means the header is sent
and properly parsed. The phone may not actually generate the response.
1xx Responses - Provisional
Response Supported Notes
100 Trying Yes
180 Ringing Yes
181 Call Is Being Forwarded No
182 Queued No
183 Session Progress Yes
B-6
2xx Responses - Success
200 OK Yes
202 Accepted Yes In REFER transfer.
3xx Responses - Redirection
300 Multiple Choices Yes
301 Moved Permanently Yes
302 Moved Temporarily Yes
305 Use Proxy No
380 Alternative Service No
4xx Responses - Request Failure
Note All 4xx responses for which the phone does not provide specific support will be
treated the same as 400 Bad Request.
400 Bad Request Yes
401 Unauthorized Yes
402 Payment Required No
403 Forbidden No
404 Not Found Yes
405 Method Not Allowed Yes
406 Not Acceptable No
407 Proxy Authentication Required Yes
408 Request Timeout No
410 Gone No
413 Request Entity Too Large No
414 Request-URI Too Long No
B-7
415 Unsupported Media Type Yes
416 Unsupported URI Scheme No
420 Bad Extension No
421 Extension Required No
423 Interval Too Brief No
480 Temporarily Unavailable Yes
481 Call/Transaction Does Not Exist Yes
482 Loop Detected Yes
483 Too Many Hops No
484 Address Incomplete Yes
485 Ambiguous No
486 Busy Here Yes
487 Request Terminated Yes
488 Not Acceptable Here Yes
491 Request Pending No
493 Undecipherable No
5xx Responses - Server Failure
500 Server Internal Error Yes
501 Not Implemented Yes
502 Bad Gateway No
503 Service Unavailable No
504 Server Time-out No
505 Version Not Supported No
513 Message Too Large No
B-8
6xx Responses - Global Failure
600 Busy Everywhere No
603 Decline Yes
604 Does Not Exist Anywhere No
606 Not Acceptable No
Hold Implementation
The phone supports both currently accepted means of signaling hold.
The first method, no longer recommended due in part to the RTCP problems
associated with it, is to set the “c” destination addresses for the media streams
in the SDP to zero, for example, c=0.0.0.0.
The second, and preferred, method is to signal the media directions with the
“a” SDP media attributes sendonly, recvonly, inactive, or sendrecv. The hold
signaling method used by the phone is configurable (refer to SIP <SIP/>on
page A-11), but both methods are supported when signaled by the remote end
point.
Note Even if the phone is set to use c=0.0.0.0, it will not do so if it gets any sendrecv,
sendonly, or inactive from the server. These flags will cause it to revert to the other
hold method.
Reliability of Provisional Responses

The phone fully supports RFC 3262 - Reliability of Provisional Responses.
Transfer
The phone supports transfer using the REFER method specified in
draft-ietf-sip-cc-transfer-05 and RFC 3515.
Third Party Call Control

The phone supports the delayed media negotiations (INVITE without SDP)
associated with third party call control applications.
When used with an appropriate server, the User Agent Computer Supported
Telecommunications Applications (uaCSTA) feature on the phone may be
utilized for remote control of the phone from computer applications such as
Microsoft Office Communicator.
B-9
The phone is compliant with “Using CSTA for SIP Phone User Agents
(uaCSTA), ECMA TR/087” for the Answer Call, Hold Call, and Retrieve Call
functions and “Services for Computer Supported Telecommunications
Applications Phase III”, ECMA – 269 for the Conference Call function.
This feature is enabled by configuration parameters described in SIP <SIP/>
on page A-11 and Registration <reg/> on page A-134 and needs to be
activated by a feature application key.
SIP for Instant Messaging and Presence Leveraging Extensions

The phone is compatible with the Presence and Instant Messaging features of
Microsoft Windows Messenger 5.1. In a future release, support for the
Presence and Instant Message recommendations in the SIP Instant Messaging
and Presence Leveraging Extensions (SIMPLE) proposals will be provided by
the following Internet drafts or their successors:
• draft-ietf-simple-cpim-mapping-01
• draft-ietf-simple-presence-07
• draft-ietf-simple-presencelist-package-00
• draft-ietf-simple-winfo-format-02
• draft-ietf-simple-winfo-package-02
Shared Call Appearance Signaling

A shared line is an address of record managed by a call server. The server
allows multiple end points to register locations against the address of record.
The phone supports shared call appearances (SCA) using the
SUBSCRIBE-NOTIFY method in the “SIP Specific Event Notification”
framework (RFC 3265). The events used are:
• “call-info” for call appearance state notification
• “line-seize for the phone to ask to seize the line
Bridged Line Appearance Signaling

A bridged line is an address of record managed by a server. The server allows
multiple end points to register locations against the address of record.
The phone supports bridged line appearances (BLA) using the
SUBSCRIBE-NOTIFY method in the “SIP Specific Event Notification”
framework (RFC 3265). The events used are:
• “dialog” for bridged line appearance subscribe and notify
B - 10
C
Miscellaneous Administrative Tasks
This appendix provides information required by varied aspects of the Session

Initiation Protocol (SIP) application. This includes:
• Trusted Certificate Authority List
• Encrypting Configuration Files
• Adding a Background Logo
• BootROM/SIP Application Dependencies
• Supported SoundStation IP 7000 / Polycom HDX Software

Interoperability
• Multiple Key Combinations
• Default Feature Key Layouts
• Internal Key Functions
• Assigning a VLAN ID Using DHCP
• Parsing Vendor ID Information
• Product, Model, and Part Number Mapping
• Disabling PC Ethernet Port
• Modifying Phone’s Configuration Using the Web Interface
• Capturing Phone’s Current Screen
• LLDP and Supported TLVs
Trusted Certificate Authority List

The following certificate authorities are trusted by the phone by default:
• ABAecom (sub., Am. Bankers Assn.) Root CA
• ANX Network CA by DST
C-1
• American Express CA
• American Express Global CA
• BelSign Object Publishing CA
• BelSign Secure Server CA
• Deutsche Telekom AG Root CA
• Digital Signature Trust Co. Global CA 1
• Entrust Worldwide by DST
• Entrust.net Premium 2048 Secure Server CA
• Entrust.net Secure Personal CA
• Entrust.net Secure Server CA
• Equifax Premium CA
• Equifax Secure CA
• Equifax Secure eBusiness CA 1
• Equifax Secure eBusiness CA 2
• Equifax Secure Global eBusiness CA 1
• GeoTrust Primary Certification Authority
• GeoTrust Global CA
• GeoTrust Global CA 2
• GeoTrust Universal CA
• GeoTrust Universal CA 2
• GTE CyberTrust Global Root
• GTE CyberTrust Japan Root CA
• GTE CyberTrust Japan Secure Server CA
• GTE CyberTrust Root 2
C-2
• GTE CyberTrust Root CA
• GlobalSign Partners CA
• GlobalSign Primary Class 1 CA
• GlobalSign Root CA
• National Retail Federation by DST
• TC TrustCenter, Germany, Class 1 CA
• Thawte Personal Basic CA
• Thawte Personal Freemail CA
• Thawte Personal Premium CA
• Thawte Premium Server CA
• Thawte Server CA
• Thawte Universal CA Root
• UPS Document Exchange by DST
• ValiCert Class 1 VA
• VeriSign Class 4 Primary CA
• Verisign Class 1 Public Primary Certification Authority
• Verisign Class 1 Public Primary Certification Authority - G2
C-3
• Verisign/RSA Commercial CA
• Verisign/RSA Secure Server CA
Polycom endeavors to maintain a built-in list of the most commonly used CA

Certificates. Due to memory constraints, we cannot keep as thorough a list as some
other applications (for example, browsers). If you are using a certificate from a
commercial Certificate Authority not in the list above, you may submit a Feature
Request for Polycom to add your CA to the trusted list by visiting
https://jira.polycom.com:8443//secure/CreateIssue!default.jspa?os_username=jirag
uest&os_password=polycom. At this point, you can use the Custom Certificate
method to load your particular CA certificate into the phone (refer to “Technical
Bulletin 17877: using Custom Certificates on SoundPoint IP Phones“ at
tins_pub.html).
Encrypting Configuration Files

The phone can recognize encrypted files, which it downloads from the
provisioning server and it can encrypt files before uploading them to the
provisioning server. There must be an encryption key on the phone to perform
these operations. Configuration files (excluding the master configuration file),
contact directories, and configuration override files can be encrypted.
A separate SDK, with a readme file, is provided to facilitate key generation and
configuration file encryption and decrypt on a UNIX or Linux server. The
utility is distributed as source code that runs under the UNIX operating
system. For more information, contact Polycom Technical Support.
A key is generated by the utility and must be downloaded to the phone so that
it can decrypt the files that were encrypted on the server. The
device.sec.configEncryption.key configuration file parameter is used to
set the key on the phone. The utility generates a random key and the
encryption is Advanced Encryption Standard (AES) 128 in Cipher Block
Chaining (CBC) mode. An example key would look like this:
Crypt=1;KeyDesc=companyNameKey1;Key=06a9214036b8a15b512e03d534120006;
If the phone doesn't have a key, it must be downloaded to the phone in plain
text (a potential security hole if not using HTTPS). If the phone already has a
key, a new key can be downloaded to the phone encrypted using the old key
(refer to Changing the Key on the Phone on page C-6). At a later date, new
phones from the factory will have a key pre-loaded in them. This key will be
changed at regular intervals to enhance security
C-4
It is recommended that all keys have unique descriptive strings in order to

allow simple identification of which key was used to encrypt a file. This makes
provisioning server management easier.
After encrypting a configuration file, it is useful to rename the file to avoid
confusing it with the original version, for example rename sip.cfg to sip.enc.
However, the directory and override filenames cannot be changed in this
manner.
You can check whether an encrypted file is the same as an unencrypted file by:
1. Run the configFileEncrypt utility on the unencrypted file with the "-d"
option. This shows the "digest" field.
2. Look at the encrypted file using WordPad and check the first line that
shows a "Digest=…." field. If the two fields are the same, then the
encrypted and unencrypted file are the same.
Note If a phone downloads an encrypted file that it cannot decrypt, the action is logged,
an error message displays, and the phone reboots. The phone will continue to do
this until the provisioning server provides an encrypted file that can be read, an
unencrypted file, or the file is removed from the master configuration file list.
Note Encrypted configuration files can only be decrypted on the SoundPoint IP 32x/33x,
430, 450, 550, 560, 650, and 670, the SoundStation IP 6000 and 7000 phones, and
the Polycom VVX 1500 phones.
The master configuration file cannot be encrypted on the provisioning server. This
file is downloaded by the BootROM that does not recognize encrypted files. For
more information, refer to Master Configuration Files on page 2-5.
The following configuration file changes are required to modify this feature:
Central Configuration File: sip.cfg Specify the phone-specific contact directory and the
(provisioning server) phone-specific configuration override file.
For more information, refer to Encryption
<encryption/> on page A-109.
Configuration file: Change the encryption key.

<device>.cfg For more information, refer to Flash Parameter
C-5
Changing the Key on the Phone

For security purposes, it may be desirable to change the key on the phones and
the server from time to time.
To change a key:
1. Put the new key into a configuration file that is in the list of files
downloaded by the phone (specified in 000000000000.cfg or <Ethernet
address>.cfg).
Use the device.sec.configEncryption.key parameter to specify the
new key.
2. Manually reboot the phone so that it will download the new key. The
phone will automatically reboot a second time to use the new key.
At this point, the phone expects all encrypted configuration files on the
provisioning server to use the new key and it will continue to reboot until
this is the case. The files on the server must be updated to the new key or
they must be made available in unencrypted format. Updating to the new
key requires decrypting the file with the old key, then encrypting it with
the new key.
Note that configuration files, contact directory files and configuration
override files may all need to be updated if they were already encrypted.
In the case of configuration override files, they can be deleted from the
provisioning server so that the phone will replace them when it
successfully boots.
Adding a Background Logo

Note Background logos are not supported on the Polycom VVX 1500 phone.
This section provides instructions on how to add a background logo to all

SoundPoint IP phones in your organization. You must be running at least
BootROM 2.x.x and SIP 1.x.x. One bitmap file is required for each model.
Model Width Height Color Depth
IP 32x/33x 102 23 monochrome
IP 430 94 23 monochrome
IP 450 256 116 4-bit grayscale or

monochrome
IP 550/560/650 209 109 4-bit grayscale or

monochrome
C-6
Model Width Height Color Depth
IP 670 209 109 12-bit color

monochrome

monochrome
Logos smaller than described in the table above are acceptable, but larger
logos may be truncated or interfere with other areas of the user interface.
RGB Values
Color RGB Values (Decimal) (Hexadecimal)
Black 0,0,0 00,00,00
Dark Gray 96,96,96 60,60,60
Light Gray 160,160,160 A0,A0,A0
White 255,255,255 FF,FF,FF
The SoundPoint IP 450/550/560/650 phone support a 4-bit grayscale, which

is a smooth gradient from black (0, 0, 0) to white (FF, FF, FF).
The SoundPoint IP 670 phone support a 12-bit color scale from black (0, 0, 0) to
white (FFFF, FFFF, FFFF).
The SoundStation IP 6000 phone is the same as the IP 7000.
The SoundStation IP 7000 phone supports a 32-bit grayscale, which is a smooth
gradient from black (0, 0, 0) to white (FF, FF, FF).

In the <bitmaps> section of sip.cfg, find the end of each model’s bitmap list
and add your bitmap to the end; do not include the .bmp extension.
Model Associate Parameter
IP 32x/33x bitmap.IP_330.68.name
IP 430 bitmap.IP_400.61.name
IP 550, 560, 650, 670 bitmap.IP_600.83.name
C-7
For example:
<bitmaps>
<IP_330 … bitmap.IP_330.68.name="logo-330" />
</bitmaps>
Next, enable the idle display feature and modify the idle display “animation”
for each model to point to your bitmap (again without the .bmp extension):
<indicators ind.idleDisplay.enabled="1">
<Animations>
<IP_330>
…
<IDLE_DISPLAY ind.anim.IP_330.30.frame.1.bitmap="logo-330"
ind.anim.IP_330.30.frame.1.duration="0"/>
…
</IP_330>
<IP_400>
…
…
</IP_400>
<IP_450>
…
…
</IP_450>
<IP_600>
…
…
</IP_600>
<IP_4000>
…
…
</IP_4000>
<IP_7000>
…
…
C-8
</IP_7000>
</Animations>
…
</indicators>
BootROM/SIP Application Dependencies

Not withstanding the hardware backward compatibility mandate, there have
been times throughout the life of the SoundPoint IP / SoundStation IP / VVX
phones where certain dependencies on specific BootROM and application
versions have been necessitated.
This table summarizes some the major dependencies that you are likely to
encounter:
Model BootROM SIP Application
IP 320/330 3.2.3B or later 2.1.1 or later
IP 321/331 4.1.3 or later 3.1.3C or later
IP 335 4.2.0B or later 4.1.2B or later
IP 430 3.1.3C or later 1.6.6 or later
IP 450 4.1.2 or later 3.1.0C or later
IP 5501 3.2.2B or later 2.1 or later
IP 5601 4.0.1 or later 2.2.2 or later
IP 650/EM1 3.2.2B or later 2.0.3B or later
IP 650/BEM 4.0.1 or later 2.2.2 or later
IP 670/CEM 4.1.1 or later 3.0.3 or later
VVX 1500 4.1.2 or later 3.1.2B or later
Note 1. SoundPoint IP 550, 560 and 650 phones manufactured as of February 2009
have additional BootROM/SIP application dependencies. For more
information, refer to “Technical Bulletin TB 46440: Notice of Product Shipping
Configuration Change” at
http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletin
s_pub.html .
2. If the SoundStation IP 7000 is connected to a Polycom HDX system, the
BootROM must be 4.1.2 or later.
C-9
Migration Dependencies
In addition to the BootROM and application dependencies, there are certain
restrictions with regard to upgrading or downgrading from one BootROM
release to another BootROM release. These restrictions are typically caused by
the addition of features that change the way BootROM provisioning is done,
so the older version become incompatible.
There is always a way to move forward with BootROM releases, although it
may be a two or three step procedure sometimes, but there are cases where it
is impossible to move backward. Make special note of these cases before
upgrading.
For the latest information, refer to the latest Release Notes.
Supported SoundStation IP 7000 / Polycom HDX Software

Interoperability
To operate your SoundStation IP 7000 phone in this environment, Polycom
recommends that you run the following combination of software on the phone
and the HDX system:
• SoundStation IP7000 running SIP 3.2.1 and BootROM 4.2.0
• Polycom HDX systems running HDX Version 2.5.0.2_29-3382 (hot fix 29)
The following table shows all supported software for the SoundStation IP 7000
and Polycom HDX systems.
For Polycom HDX 4000 • Recommended: SIP 3.2.1/BootROM 4.2.0 in

HDX 8000 combination with HDX 2.5.0.2_29-3382 (hot fix 29)
HDX 9000 • Compatible: SIP 3.1.1RevB / BootROM 4.1.2
through SIP 3.1.3RevC / BootROM 4.1.3 in
combination with HDX 2.5 through 2.5.0.5

combination with HDX 2.5.0.2_29-3382 (hot fix 29)
• Compatible: SIP 3.1.2RevC / BootROM 4.1.2
through SIP 3.1.3RevC / BootROM 4.1.3 in
combination with HDX 2.5 through 2.5.0.5

combination with HDX 2.5.0.7 planned for Q4 2009
C - 10
Multiple Key Combinations

On SoundPoint IP and SoundStation IP phones, certain multiple key
combinations can be used to reboot the phone and restore factory defaults.
For other methods for resetting and rebooting your SoundPoint IP,
SoundStation IP, or Polycom VVX phones, refer to “Quick Tip 18298: Resetting
and Rebooting SoundPoint IP / SoundStation IP / VVX Phones” at
Rebooting the Phone

For the key combination, press and hold certain key combinations (depending
on the phone model) simultaneously until a confirmation tone is heard or for
about three seconds:
• IP 32x/33x: Volume-, Volume+, Hold, and Hands-free
• IP 430: Volume-, Volume+, Hold, and Messages
• IP 450, 550, 560, 600, 601, and 650, and 670: Volume-, Volume+, Mute, and
Messages
• IP 6000: *, #, Volume+, and Select
• IP 7000: *, #, Volume-, and Volume+
• VVX 1500: Delete, Volume-, Volume+, and Select
Note As of SIP 3.2.0, users can restart their phones by pressing the Menu key, and then
selecting Settings > Basic > Restart Phone. Any new BootROM and SIP
applications will be downloaded to the phone as a result of this restart.
Restoring Factory Defaults

on the phone model) simultaneously during the countdown process in the
BootROM until the password prompt appears:
• IP 450, 550, 600, 601, and 650, and 670 and VVX 1500: 4, 6, 8 and * dial pad
keys
• IP 32x/33x, 430, 560, 7000: 1, 3, 5, and 7 dial pad keys
• IP 6000: 6, 8 and * dial pad keys

Enter the administrator password to initiate the reset. Resetting to factory
defaults will also reset the administrator password (factory default password
is 456). Polycom recommends that you change the administrative password
from the default value.
C - 11
Uploading Log Files

on the phone model) simultaneously until a confirmation tone is heard or for
about three seconds:
• IP 32x/33x: Menu, Dial, and the two Line keys
• IP 430, 450, 550, 560, 600, 601, 650, 670, and 7000 and VVX 1500: Up, Down,
Left, and Right arrow keys
• IP 6000: Menu, Exit, Off-hook/Hands-free, Redial
C - 12
Default Feature Key Layouts

The following figures and tables show the default SIP key layouts for the
SoundPoint IP 32x/33x, 430, 450, 550, 560, 650, and 670, SoundStation IP 6000
and 7000, and Polycom VVX 1500 models.
SoundPoint IP 320/321/330/331/335
31 13 7
14
Menu Line 1
32 15 8
33 9
Dial Line 2
34 10
16
ABC DEF
1 6 2 1 325
Hold
GHI JKL MNO 19
4 5
5 2 6 26
PQRS TUV WXYZ 20
7 4 8 3 9 27
30 OPER 21
29
0 28 22
24 23
Key ID
Key ID Function Key ID Function Key ID Function Key ID Function
1 Dialpad2 12 n/a 23 VolUp 34 Menu
2 Dialpad5 13 SoftKey2 24 VolDown 35 n/a
3 Dialpad8 14 ArrowUp 25 Dialpad3 36 n/a
4 Dialpad7 15 Select 26 Dialpad6 37 n/a
5 Dialpad4 16 ArrowDown 27 Dialpad9 38 n/a
6 Dialpad1 17 n/a 28 Dialpad0 39 n/a
7 SoftKey3 18 n/a 29 DialpadStar 40 n/a
8 Line1 19 Hold 30 MicMute 41 n/a
9 ArrowRight 20 Headset 31 SoftKey1 42 n/a
10 Line2 21 Handsfree 32 Dial
11 n/a 22 DialpadPound 33 ArrowLeft
C - 13
SoundPoint IP 430
1 Line1 12 DialpadPound 23 Messages 34 Softkey3
2 Line2 13 Dialpad9 24 n/a 35 Handsfree
3 n/a 14 Dialpad8 25 SoftKey4 36 n/a
4 ArrowUp 15 Dialpad7 26 Headset 37 n/a
5 Hold 16 Dialpad4 27 SoftKey2 38 n/a
7 Redial 18 Dialpad6 29 ArrowDown 40 n/a
8 VolUp 19 Dialpad3 30 Select 41 n/a
9 VolDown 20 Dialpad2 31 ArrowLeft 42 n/a
10 DialpadStar 21 Dialpad1 32 Menu
11 Dialpad0 22 ArrowRight 33 MicMute
C - 14
SoundPoint IP 450
1
2
3
4
30
28 27 34 25 31 22
29
5 35
26
32
21 20 19
23
16 17 18
7
15 14 13
33
10 11 12
9 8
Key ID
1 Line1 12 DialpadPound 23 Messages 34 SoftKey3
2 Line2 13 Dialpad9 24 n/a 35 Handsfree
3 Line3 14 Dialpad8 25 Softkey4 36 n/a
4 ArrowUp 15 Dialpad7 26 Headset 37 n/a
5 Hold 16 Dialpad4 27 SoftKey2 38 n/a
7 Redial 18 Dialpad6 29 ArrowDown 40 n/a
8 VolUp 19 Dialpad3 30 Select 41 n/a
9 VolDown 20 Dialpad2 31 ArrowLeft 42 n/a
10 DialpadStar 21 Dialpad1 32 Menu
11 Dialpad0 22 ArrowRight 33 MicMute
C - 15
SoundPoint IP 550/560/650/670
34 1
33
Sel
35 2 5 4
41
42 3
31
Del
6
Directories Menu
30 28 27 26 25 7
Services Messages
29
1 24 2 23 3
ABC DEF
8
Conference 22 Do Not Disturb
ABC DEF
32 9
4 19 5 20 6
GHI JKL MNO
Transfer
37 GHI JKL MNO 21 10
77 88 99 16
PQRS TUV WXYZ
Redial
Hold
36 18 TUV 17 39 38
PQRS WXYZ
Hold
0
OPER
40 * 15 OPER
14
#
13
12 11
Key ID
Note The SoundPoint IP 550 and 560 has have only the top four lines keys. Key IDs 31
and 42 are not used on SoundPoint IP 550 and 560 phones.
1 ArrowUp 12 VolDown 23 Dialpad2 34 Line1
2 ArrowLeft 13 DialpadPound 24 Dialpad1 35 Line3
3 ArrowDown 14 Dialpad0 25 SoftKey4 36 Redial
4 ArrowRight 15 DialpadStar 26 SoftKey3 37 Transfer
5 Select 16 Dialpad9 27 SoftKey2 38 Headset
6 Delete 17 Dialpad8 28 SoftKey1 39 Handsfree
7 Menu 18 Dialpad7 29 Applications 40 Hold
8 Messages 19 Dialpad4 30 Directories 41 Line4
9 DoNotDisturb 20 Dialpad5 31 Line6 42 Line5
10 MicMute 21 Dialpad6 32 Conference
11 VolUp 22 Dialpad3 33 Line2
C - 16
SoundStation IP 6000
27 17
23
26 11
28 29 25
1 2 3 4
5
7 8 9 10
22 13 14 15 16
19 20 21
Key ID
1 Dialpad1 12 n/a 23 Select 34 n/a
2 Dialpad2 13 Dialpad7 24 n/a 35 n/a
3 Dialpad3 14 Dialpad8 25 SoftKey3 36 n/a
4 VolUp 15 Dialpad9 26 Exit 37 n/a
5 Handsfree 16 MicMute 27 Menu 38 n/a
6 n/a 17 ArrowUp 28 SoftKey1 39 n/a
7 Dialpad4 18 n/a 29 SoftKey2 40 n/a
8 Dialpad5 19 DialpadStar 30 n/a 41 n/a
9 Dialpad6 20 Dialpad0 31 n/a 42 n/a
10 VolDown 21 DialpadPound 32 n/a
11 ArrowDown 22 Redial 33 n/a
C - 17
SoundStation IP 7000
1 7 13 19
26
6 14 20 30
8
9 15 21
3 27
10 16 22
4 28
11 17 23
5 29
12 18 24
Key ID
1 SoftKey1 12 DialpadStar 23 Dialpad9 34 n/a
2 ArrowUp 13 SoftKey3 24 DialpadPound 35 n/a
3 Menu 14 ArrowLeft 25 n/a 36 n/a
4 Conference 15 Dialpad2 26 Select 37 n/a
5 Redial 16 Dialpad5 27 VolUp 38 n/a
6 Handsfree 17 Dialpad8 28 VolDown 39 n/a
7 SoftKey2 18 Dialpad0 29 MicMute 40 n/a
8 ArrowDown 19 SoftKey4 30 Release 41 n/a
9 Dialpad1 20 ArrowRight 31 n/a 42 n/a
10 Dialpad4 21 Dialpad3 32 n/a
11 Dialpad7 22 Dialpad6 33 n/a
C - 18
Polycom VVX 1500
37 25 13 1
8
2 4
39 33 27 3
20 14
40 34 28 21 15
5
41 35 29 22 16
24 18
42 36 30 23 17 12
Key ID
1 Messages 12 MicMute 23 Headset 34 Dialpad5
2 ArrowLeft 13 Directories 24 n/a 35 Dialpad8
3 Select 14 Redial 25 Menu 36 Dialpad0
4 ArrowRight 15 Conference 26 n/a 37 Applications
5 Delete 16 DoNotDisturb 27 Dialpad3 38 n/a
6 n/a 17 Handsfree 28 Dialpad6 39 Dialpad1
7 n/a 18 VolUp 29 Dialpad9 40 Dialpad4
8 ArrowUp 19 n/a 30 DialpadPound 41 Dialpad7
9 ArrowDown 20 Video 31 n/a 42 DialpadStar
10 n/a 21 Transfer 32 n/a
11 n/a 22 Hold 33 Dialpad2
Internal Key Functions

A complete list of internal key functions for enhanced feature keys and hard
key mappings is shown in the following table.
The following guidelines should be noted:
• The Label value is case sensitive.
• Some functions are dependent on call state. Generally, if the soft key
appears on a call screen, the soft key function is executable. There are some
exceptions on the SoundPoint IP 32x/33x phone (because it does not
display as many soft keys).
C - 19
• On the SoundPoint IP 32x/33x phone, CallPickup and ParkedPickup refer

to the same function. On other phones, CallPickup refers to the soft key
function that provides the menu with separate soft keys for parked
pickup, directed pickup, and group pickup.
• Some functions depend on the feature being enabled. For example,

BuddyStatus and MyStatus require the presence feature to be enabled.
• Hard key remappings do not require the Enhanced Feature key feature to
be enabled. This include the SpeedDial function on older platforms. On
newer platforms, use line key functions.
• The table below shows only Line1 to Line6 functions. For the SoundPoint
IP 650 and 670 phones with attached Expansion Modules, Line7 to Line48
functions are also supported.
Label Function Notes
ACDAvailable ACDAvailableFromIdle
ACDLogin ACDLoginLogout
ACDLogout ACDLoginLogout
ACDUnavailable ACDAvailableFromIdle
Answer Answer Call screen only
Applications Main Browser
ArrowDown ArrowDown
ArrowLeft ArrowLeft
ArrowRight ArrowRight
ArrowUp ArrowUp
BargeIn BargInShowAppearances, BargeIn Call screen only
BuddyStatus Buddy Status
Callers Callers
CallList Call Lists
CallPark ParkEntry Call screen only
CallPickup CallPickupEntry Call screen only
Conference ConferenceCall Call screen only
Delete Delete
Dialpad0 Dialpad0
Dialpad1 Dialpad1
Dialpad2 Dialpad2
C - 20
Dialpad3 Dialpad3
Dialpad4 Dialpad4
Dialpad5 Dialpad5
Dialpad6 Dialpad6
Dialpad7 Dialpad7
Dialpad8 Dialpad8
Dialpad9 Dialpad9
DialpadPound DialpadPound
DialpadStar DialpadStar
DialpadURL Dialname Call screen only
DirectedPiclup DirectedPickup Call screen only
Directories Directories
Divert Forward
DoNotDisturb Do Not Disturb menu
Exit Exist existing menu Menu only
GroupPickup GroupPickup
Handsfree Handsfree
Headset Headset Desktop phones only
Hold Toggle Hold
Join Join Call screen only
LCR LastCallReturn
Line1 Line Key 1
Line2 Line Key 2
Line3 Line Key 3
Line4 Line Key 4
Line5 Line Key 5
Line6 Line Key 6
ListenMode Turn on speaker to listen only
Menu Menu
Messages Messages menu
C - 21
MicMute MicMute
MyStatus MyStatus
NewCall NewCall Call screen only
Null Do nothing
Offline Offline for presence
QuickSetup Quick Setup feature Call screen only
EnterRecord enterCallRecord Call screen only
Redial Redial Call screen only
Release EndCall or Cancel hot dial SoundStation IP 7000 only
ParkedPickup ParkedPickup Call screen only
Select Select
ServerACDAgentAvailable serverACDAgentAvailable
ServerACDAgentUnavailable serverACDAgentUnavailable
ServerACDSignIn serverACDSignIn
ServerACDSignOut serverACDSignOut
Setup Settings menu
Silence RingerSilence Call screen only
SoftKey1 SoftKey1
SoftKey2 SoftKey2
SoftKey3 SoftKey3
SoftKey4 SoftKey4
SpeedDial SpeedDial
Split Split Call screen only
Transfer Transfer Call screen only
Video Video Polycom VVX 1500 only
VolDown VolDown
VolUp VolUp
C - 22
Assigning a VLAN ID Using DHCP
To assign a VLAN ID to a phone using DHCP:

>> In the DHCP menu of the Main setup menu, set VLAN Discovery to
“Fixed” or “Custom”.
When set to “Fixed”, the phone will examine DHCP options 128,144, 157
and 191 (in that order) for a valid DVD string.
When set to "Custom", the value set in "VLAN ID Option" will be
examined for a valid DVD string.
DVD string in the DHCP option must meet the following conditions to be
valid:
— Must start with ?VLAN-A=? (case-sensitive)
— Must contain at least one valid ID
— VLAN IDs range from 0 to 4095
— Each VLAN ID must be separated by a ?+? character
— The string must be terminated by a ?;?
— All characters after the ?;? will be ignored
— There must be no white space before the ?;?
— VLAN IDs may be decimal, hex, or octal
For example:
The following DVD strings will result in the phone using VLAN 10:
VLAN-A=10;
VLAN-A=0x0a;
VLAN-A=012;
Note If a VLAN tag is assigned by CDP, DHCP VLAN tags will be ignored.
C - 23
The following figure shows the phone’s processing to determine if the VLAN
ID is valid:
Parsing Vendor ID Information

After the phone boots, it sends a DHCP Discover packet to the DHCP server.
This is found in the Bootstrap Protocol/option “Vendor Class Identifier”
section of the packet and includes the phone’s part number and the BootROM
version. The format of this option's data is not specified in RFC 2132, but is left
to each vendor to define its own format. To be useful, every vendor's format
must be distinguishable from every other vendor's format. To make our
format uniquely identifiable, the format follows RFC 3925, which uses the
C - 24
IANA Private Enterprise number to determine which vendor's format should

be used to decode the remaining data. The private enterprise number assigned
to Polycom is 13885 (0x0000363D).
This vendor ID information is not a character string, but an array of binary
data. The steps for parsing are as follows:
1. Check for the Polycom signature at the start of the option:
4 octet: 00 00 36 3d
2. Get the length of the entire list of sub-options:
1 octet
3. Read the field code and length of the first sub-option, 1+1 octets
4. If this is a field you want to parse, save the data.
5. Skip to the start of the next sub-option.
6. Repeat steps 3 to 5 until you have all the data or you encounter the
End-of-Suboptions code (0xFF).
For example, the following is a sample decode of a packet from an IP601:
3c 74
- Option 60, length of Option data (part of the DHCP spec.)
00 00 36 3d
- Polycom signature (always 4 octects)
6f
- Length of Polycom data
01 07 50 6f 6c 79 63 6f 6d
- sub-option 1 (company), length, "Polycom"
02 15 53 6f 75 6e 64 50 6f 69 6e 74 49 50 2d 53 50 49 50 5f 36 30 31
- sub-option 2 (part), length, "SoundPointIP-SPIP_601"
03 10 32 33 34 35 2d 31 31 36 30 35 2d 30 30 31 2c 32
- sub-option 3 (part number), length, "2345-11605-001,2"
04 1c 53 49 50 2f 54 69 70 2e 58 58 58 58 2f 30 38 2d 4a 75 6e 2d 30 37
20 31 30 3a 34 34
- sub-option 4 (Application version), length, "SIP/Tip.XXXX/08-Jun-07
10:44"
05 1d 42 52 2f 33 2e 31 2e 30 2e 58 58 58 58 2f 32 38 2d 41 70 72 2d 30
35 20 31 33 3a 33 30
- sub-option 5 (BootROM version), length, "BR/3.1.0.XXXX/28-Apr-05
13:30"
ff
- end of sub-options
For the BootROM, sub-option 4 and sub-option 5 will contain the same string.
The string is formatted as follows:
<apptype>/<buildid>/<date+time>
where:
<apptype> can be 'BR' (BootROM) or 'SIP' (SIP Application)
C - 25
Product, Model, and Part Number Mapping

In SIP 2.1.2, enhancements to the master configuration file were made to allow
you to direct phone upgrades to a software image and configuration files
based on phone model number, firmware part number, or MAC address.
The part number specific version has precedence over the model number
version, which has precedence over the original version. For example,
CONFIG_FILES_2345-11560-001=”phone1_2345-11560-001.cfg,
sip_2345-11560-001.cfg” will override
CONFIG_FILES_SPIP560=”phone1_SPIP560.cfg, sip_SPIP560.cfg”, which will
override CONFIG_FILES=”phone1.cfg, sip.cfg” for an SoundPoint IP 560.
You can also add variables to the master configuration file that are replaced
when the phone reboots. The variables include PHONE_MODEL,
PHONE_PART_NUMBER, and PHONE_MAC_ADDRESS.
The following table shows the product name, model name, and part number
mapping for SoundPoint IP, SoundStation IP, and Polycom VVX 1500 phones:
Product Name Model Name Product Part Number
SoundPoint IP 300 SPIP300 2345-11300-001
SoundPoint IP 320 SPIP320 2345-12200-002,

2345-12200-005

2345-12200-004

2345-11500-010,
2345-11500-020

2345-11500-040
C - 26
Product Name Model Name Product Part Number
SoundStation IP 4000 SSIP4000 2201-06642-001
Polycom VVX 1500 VVX1500 2345-17960-001
Disabling PC Ethernet Port

Certain SoundPoint IP phones have a PC Ethernet port. If it is unused, it can
be disabled.
The PC Ethernet port can be disabled on the SoundPoint IP 33x, 430, 450, 550,
560, 601, 650, and 670, and Polycom VVX 1500 through the menu (shown
below). The Ethernet port can also be disabled through the configuration files.
To disable the Ethernet port on a supported SoundPoint IP phone:
1. Press .
2. Select Settings > Advanced > Network Configuration > Ethernet Menu.
You must enter the administrator password to access the network
configuration. The factory default password is 456.
3. Scroll down to PC Port Mode and select Edit.
4. Select Disabled, and then press the OK soft key.
5. Press the Exit soft key.
6. Select Save Config.
The SoundPoint IP phone reboots. When the reboot is complete, the PC
Ethernet port is disabled.
Modifying Phone’s Configuration Using the Web Interface

You can make changes to the configuration files through the web interface to
the phone.
C - 27
To configure your phone through the web interface:

>> Using your chosen browser, do the following:
a To get your phone’s IP address, press the Menu key, and then
selecting Status > Platform > Phone. Scroll down to see the IP
address.
b Enter your phone’s IP address as the browser address.
A web page similar to the one shown below appears.
c Select SIP from the menu tab.

You will be prompted for the SIP username and password.
d Make the desired configuration changes.

e Scroll down to the bottom of the Servers section.
C - 28
f Select the Submit button.

Your phone will reboot.

g Select General from the menu tab.
h If you make any changes, scroll down to the bottom of the section.
i Select the Submit button.
Your phone will reboot.
C - 29
Capturing Phone’s Current Screen

You can capture the current screen on a SoundPoint IP, SoundStation IP or
Polycom VVX phone through the web interface to the phone.
To capture the phone’s current screen:

b Locate the user preference parameter.
c Change up.screenCapture.enabled to 1.
d Save the modified sip.cfg configuration file.
2. On the phone, do the following:
a Press the Menu key, and then select Settings > Basic > Preferences >
Screen Capture.
b Using the arrow keys, select Enabled, and then press the Select soft
key.
Note You need to re-enable the Screen Capture feature after every phone restart or
reboot (repeat step 2).
1. Using your chosen browser, do the following:

— To get your phone’s IP address, press the Menu key, and then select
Status > Platform > Phone. Scroll down to see the IP address.
— As the browser address, enter http://<phone’s IP
address>/captureScreen .
The current screen that is shown on the phone is shown in the browser
window. The image can be saved as a BMP or JPEG file.
LLDP and Supported TLVs

The Link Layer Discovery Protocol (LLDP) is a vendor-neutral Layer 2
protocol that allows a network device to advertise its identity and capabilities
on the local network. The protocol was formally ratified as IEEE standard
802.1AB- 2005 in May 2005. Refer to section 10.2.4.4 of the LLDP-MED
standard at
http://www.tiaonline.org/standards/technology/voip/documents/ANSI-
TIA-1057_final_for_publication.pdf .
C - 30
The LLDP feature (added in SIP 3.2.0) supports VLAN discovery and LLDP
power management, but not power negotiation. LLDP has a higher priority
than CDP and DHCP VLAN discovery.
The following Type Length Values (TLVs) are supported:
• Mandatory
— Chassis ID—Must be first TLV
— Port ID—Must be second TLV
— Time-to-live—Must be third TLV, set to 120 seconds
— End-of-LLDPDU—Must be last TLV
— LLDP-MED Capabilities
— LLDP-MED Network Policy—VLAN, L2 QoS, L3 QoS
— LLDP-MED Extended Power-Via-MDI TLV—Power Type, Power
Source, Power Priority, Power Value
• Optional
— Port Description
— System Name—Administrator assigned name
— System Description—Includes device type, phone number, hardware
version, and software version
— System Capabilities—Set as "Telephone" capability
— MAC / PHY config status—Detects duplex mismatch
— Management Address—Used for network discovery
— LLDP-MED Location Identification—Location data formats:
Co-ordinate, Civic Address, ECS ELIN
— LLDP-MED Inventory Management —Hardware Revision, Firmware
Revision, Software Revision, Serial Number, Manufacturer’s Name,
Model Name, Asset ID
An LLDP frame shall contain all mandatory TLVs. The frame will be
recognized as LLDP only if it contains mandatory TLVs. SoundPoint IP /
SoundStation IP / VVX phones will support LLDP frames with both
mandatory and optional TLVs. The basic structure of an LLDP frame and a
table containing all TLVs along with each field is explained in Supported TLVs
on page C-32.
C - 31
Note As per section 10.2.4.4 of the LLDP-MED standard, LLDP-MED endpoint devices
need to transmit Location Identification TLVs if they are capable of either
automatically determining their physical location by use of GPS or radio beacon or
capable of being statically configured with this information.
At present, the SoundPoint IP / SoundStation IP / VVX phones do not have the
capability to determine their physical location automatically or provision to a
statically configured location. Because of these limitations, the SoundPoint IP /
SoundStation IP / VVX phones will not transmit Location Identification TLV in the
LLDP frame. However, the location information from the switch is decoded and
displayed on the phone’s menu.
For more information on configuration parameters, refer to Flash Parameter

Supported TLVs
This is the basic TLV format:
TLV Type (7 bits) [0-6] TLV Length (9 bits) TLV Information (0-511
[7-15] bytes)
The following is a list of supported TLVs:
Org. Version
Type Length Unique
(7 bits) (9 bits) Type Code Sub-
No Name [0-6] [7-15] Length (3 bytes) Type Information
1 Chassis-Id1 1 6 0x0206 - 5 IP address of phone (4

bytes)
Note: 0.0.0.0 is sent until
the phone has a valid IP
address.
2 Port-Id1 2 7 0x0407 - 3 MAC address of phone (6

bytes)
3 TTL 3 2 0x0602 - - TTL value is 120/0 sec
4 Port 4 1 0x0801 - - Port description 1

description
5 System 5 min len > - - - Refer to System Names on

name 0, max len page C-37
<= 255
C - 32
Org. Version
Type Length Unique
6 System 6 min len > - - - Manufacturer’s name -

description 0, max len “Polycom”; Refer to Model
<= 255 Names on page C-37;
Hardware version;
Application version;
BootROM version
7 Capabilities 7 4 0x0e04 - - System Capabilities:

Telephone and Bridge if
the phone has PC port
support and it is not
disabled.
Enabled Capabilities:
Telephone and Bridge if
phone has PC port
support, it is not disabled
and PC port is connected
to PC.
Note:
PC port supported Phones:
IP 330, IP 331, IP 335, IP
430, IP 450, IP 550, IP
560, IP 650, and IP 670.
PC port not supported
phones:
IP6000, IP7000, IP320,
and IP321.
8 Manageme 8 12 0x100c - - Address String Len - 5,

nt Address IPV4 subtype, IP address,
Interface subtype -
“Unknown”, Interface
number - “0”, ODI string
Len - “0”
9 IEEE 802.3 127 9 0xfe09 0x00120f 1 Auto Negotiation

MAC/PHY Supported - “1”,
config/statu enabled/disabled, Refer to
s1 PMD Advertise and
Operational MAU on page
C-38
C - 33
Org. Version
Type Length Unique
10 LLDP-MED 127 7 0xfe07 0x0012bb 1 Capabilities - 0x33

capabilities (LLDP-Med capabilities,
Network policy, Extended
Power Via MDI-PD,
Inventory)
Class Type III
Note: Once support for
configuring location
Identification information is
locally available:
Capabilities - 0x37
(LLDP-Med capabilities,
Network policy, Location
Identification, Extended
Power Via MDI-PD,
Inventory)
Class Type III
11 LLDP-MED 127 8 0xfe08 0x0012bb 2 ApplicationType: Voice (1),

network Policy:
policy2 (Unknown(=1)/Defined(=0)
Unknown, if phone is in
booting stage or if switch
doesn't support network
policy TLV.
Defined, if phone is
operational stage and
Networkpolicy TLV is
received from the switch.),
Tagged/Untagged, VlanId,
L2 priority and DSCP
C - 34
Org. Version
Type Length Unique
12 LLDP-MED 127 8 0xfe08 0x0012bb 2 ApplicationType: Voice

network Signaling (2), Policy:
policy TLV.
received from the
switch.),Tagged/Untagged,
VlanId, L2 priority and
DSCP.
Note: Voice signaling TLV
is sent only if it contains
configuration parameters
that are different from
voice parameters.
13 LLDP-MED 127 8 0xfe08 0x0012bb 2 ApplicationType: Video

network Conferencing (6),Policy:
policy TLV.
received from the
switch.),Tagged/Untagged,
VlanId, L2 priority and
DSCP.
Note: Video Conferencing
TLV is sent only from Video
capable phones (currently
Polycom VVX 1500 only).
14 LLDP-MED 127 min len > - 0x0012bb 3 ELIN data format: 10 digit
location 0, max len emergency number
identificatio <= 511 configured on the switch.
n3 Civic Address: physical
address data such as city,
street number, and building
information.
C - 35
Org. Version
Type Length Unique
15 Extended 127 7 0xfe07 0x0012bb 4 PowerType -PD device

power via PowerSource-PSE&local
MDI
Power Priority -Unknown
PowerValue - Refer to
Power Values on page
C-39
16 LLDP-MED 127 min len > - 0x0012bb 5 Hardware part number and
inventory 0, max len revision
hardware <= 32
revision
17 LLDP-MED 127 min len > - 0x0012bb 6 BootROM revision

inventory 0, max len
firmware <= 32
revision
18 LLDP-MED 127 min len > - 0x0012bb 7 Application (SIP) revision

inventory 0, max len
software <= 32
revision
19 LLDP-MED 127 min len > - 0x0012bb 8 MAC Address (ASCII

inventory 0, max len string)
serial <= 32
number
20 LLDP-MED 127 11 0xfe0b 0x0012bb 9 Polycom

inventory
manufactur
er name
21 LLDP-MED 127 min len > - 0x0012bb 10 Refer to Model Names on

inventory 0, max len page C-37
model <= 32
name
22 LLDP-MED 127 4 0xfe08 0x0012bb 11 Empty (Zero length string)

inventory
asset ID
23 End of 0 0 0x0000 - - -
LLDP DU
C - 36
Note 1. For other subtypes, refer to IEEE 802.1AB, March 2005 at

http://www.ieee802.org/1/pages/802.1ab.html .
2. For other application types, refer to TIA Standards 1057, April 2006 at
http://tia.nufu.eu/std/ANSI|TIA-1057 .
3. At this time, this TLV is not sent by the phone.
System Names
Model System Name
IP 320 Polycom SoundPoint IP 320
IP 6000 Polycom SoundStation IP 6000
IP 7000 Polycom SoundStation IP 7000
VVX 1500 Polycom VVX 1500
Model Names
Model Model Name
IP 320 SoundPointIP-SPIP_320
C - 37
Model Model Name
IP 6000 SoundStationIP-SSIP_6000
IP 7000 SoundStationIP-SSIP_7000
VVX 1500 VVX-VVX_1500
PMD Advertise and Operational MAU
PMD Advertise
Mode/Speed Capability Bit Operational MAU Type
10BASE-T half duplex 1 10

mode
10BASE-T full duplex 2 11

mode

mode

mode

mode

mode
Unknown 0 0
Note By default, all phones have the PMD Advertise Capability set for 10HD, 10FD,
100HD and 100FD bits. For SoundPoint IP 560 and IP 670, and Polycom VVX 1500
phones that have Gigabit Ethernet support PMD Advertise Capability also contains
set 1000FD bit.
C - 38
Power Values
Power Value Sent in

LLDP-MED Extended
Model Power Usage (Watts) Power Via MDI TLV
IP 320/330 4.5 45
IP 321/331 4.5 45
IP 335 5.5 55
IP 430 4.5 45
IP 450 4.5 45
IP 550 6 60
IP 560 8 80
IP 650 with EM 12 120
IP 670 with EM 14 140
IP 6000 10.5 105
IP 7000 10.5 105
VVX 1500 14 140
Note By default, the power values for the SoundPoint IP 650 and 670 are sent for the
phone and the Expansion Module(s). The values are not adjusted when the
Expansion Module(s) are detached from the phone.
C - 39
C - 40
D
Third Party Software
This appendix provides the copyright statements for third party software
products that are part of the application programs that run on Polycom
SoundPoint IP, SoundStation IP, and VVX 1500 phones.
Product License Location
c-ares c-ares on page D-2
curl curl on page D-3
eXpat eXpat on page D-9
ILG JPEG IJG JPEG on page D-9
libMng libMng on page D-10
libPng libPng on page D-11
libSRTP libSRTP on page D-13
libssh2 libssh2 on page D-13
OpenLDAP OpenLDAP on page D-14
OpenSSL OpenSSL on page D-15
zlib zlib on page D-18
This appendix provides the copyright statements for third party software
products that are part of the application programs that run on Polycom VVX
1500 phones only.
BusyBox Refer to the “Polycom Voice OFFER of Source for GPL

and LGPL Software”
dhcp dhcp 4.0.0-14 on page D-3
droidfonts droidfonts on page D-5
Dropbear Dropbear on page D-4
D-1
glibc Refer to the “Polycom Voice OFFER of Source for GPL

libstdc++ Refer to the “Polycom Voice OFFER of Source for GPL

Linux kernel Refer to the “Polycom Voice OFFER of Source for GPL
module-init-tools Refer to the “Polycom Voice OFFER of Source for GPL

mtd-utils Refer to the “Polycom Voice OFFER of Source for GPL

ncurses ncurses on page D-14
pmap pmap-29092002 on page D-17
procps Refer to the “Polycom Voice OFFER of Source for GPL

tsattach Refer to the “Polycom Voice OFFER of Source for GPL

tslib Refer to the “Polycom Voice OFFER of Source for GPL

udev Refer to the “Polycom Voice OFFER of Source for GPL

Webkit Refer to the “Polycom Voice OFFER of Source for GPL

wrsv-ltt Refer to the “Polycom Voice OFFER of Source for GPL

The “Polycom Voice OFFER of Source for GPL and LGPL Software” is
available at
http://downloads.polycom.com/voice/voip/offerForSourceVoiceProducts.
html .
c-ares
Copyright 1998 by the Massachusetts Institute of Technology.
Permission to use, copy, modify, and distribute this software and its
documentation for any purpose and without fee is hereby granted, provided
that the above copyright notice appear in all copies and that both that
copyright notice and this permission notice appear in supporting
documentation, and that the name of M.I.T. not be used in advertising or
publicity pertaining to distribution of the software without specific, written
prior permission.
D-2
M.I.T. makes no representations about the suitability of this software for any
purpose. It is provided "as is" without express or implied warranty.
curl
COPYRIGHT AND PERMISSION NOTICE
Copyright (c) 1996 - 2008, Daniel Stenberg, <daniel@haxx.se>.
All rights reserved.
Permission to use, copy, modify, and distribute this software for any purpose
with or without fee is hereby granted, provided that the above copyright
notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY
KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS. IN NO
EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE
FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT
OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
Except as contained in this notice, the name of a copyright holder shall not be
used in advertising or otherwise to promote the sale, use or other dealings in
this Software without prior written authorization of the copyright holder.
dhcp 4.0.0-14
Copyright (c) 2004-2009 by Internet Systems Consortium, Inc. ("ISC")
Copyright (c) 1995-2003 by Internet Software Consortium
Permission to use, copy, modify, and distribute this software for any purpose
with or without fee is hereby granted, provided that the above copyright
notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS" AND ISC DISCLAIMS ALL
WARRANTIES WITH REGARD TO THIS SOFTWARE INCLUDING ALL
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS. IN NO
EVENT SHALL ISC BE LIABLE FOR ANY SPECIAL, DIRECT, INDIRECT,
OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES WHATSOEVER
RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION,
ARISING OUT OF OR IN CONNECTION WITH THE USE OR
PERFORMANCE OF THIS SOFTWARE.
Internet Systems Consortium, Inc.
950 Charter Street
Redwood City, CA 94063
<info@isc.org>
D-3
http://www.isc.org/
Dropbear
The majority of code is written by Matt Johnston, under the license below.
Portions of the client-mode work are (c) 2004 Mihnea Stoenescu, under the
same license:
Copyright (c) 2002-2006 Matt Johnston
Portions copyright (c) 2004 Mihnea Stoenescu
Permission is hereby granted, free of charge, to any person obtaining a copy of
this software and associated documentation files (the "Software"), to deal in
the Software without restriction, including without limitation the rights to use,
copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the
Software, and to permit persons to whom the Software is furnished to do so,
subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
=====
LibTomCrypt and LibTomMath are written by Tom St Denis, and are Public
Domain.
=====
sshpty.c is taken from OpenSSH 3.5p1,
Copyright (c) 1995 Tatu Ylonen <ylo@cs.hut.fi>, Espoo, Finland
All rights reserved
"As far as I am concerned, the code I have written for this software can be used
freely for any purpose. Any derived versions of this software must be clearly
marked as such, and if the derived work is incompatible with the protocol
description in the RFC file, it must be called by a name other than "ssh" or
"Secure Shell". "
=====
loginrec.c
loginrec.h
D-4
atomicio.h
atomicio.c
and strlcat() (included in util.c) are from OpenSSH 3.6.1p2, and are licensed
under the 2 point BSD license.
loginrec is written primarily by Andre Lucas, atomicio.c by Theo de Raadt.
strlcat() is (c) Todd C. Miller
=====
Import code in keyimport.c is modified from PuTTY's import.c, licensed as
follows:
PuTTY is copyright 1997-2003 Simon Tatham.
Portions copyright Robert de Bath, Joris van Rantwijk, Delian Delchev,
Andreas Schultz, Jeroen Massar, Wez Furlong, Nicolas Barry, Justin Bradford,
and CORE SDI S.A.
COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR
OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
droidfonts
Apache License
Version 2.0, January 2004
http://www.apache.org/licenses/
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DISTRIBUTION
1. Definitions.
"License" shall mean the terms and conditions for use, reproduction, and
distribution as defined by Sections 1 through 9 of this document.
D-5
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outstanding shares, or (iii) beneficial ownership of such entity.
"You" (or "Your") shall mean an individual or Legal Entity exercising
permissions granted by this License.
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including but not limited to software source code, documentation source, and
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"Work" shall mean the work of authorship, whether in Source or Object form,
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non-exclusive, no-charge, royalty-free, irrevocable copyright license to
D-6
reproduce, prepare Derivative Works of, publicly display, publicly perform,

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2. You must cause any modified files to carry prominent notices stating
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this License.
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END OF TERMS AND CONDITIONS
D-8
eXpat
Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd and
Clark Cooper
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
IJG JPEG
Independent JPEG Group's free JPEG software
This package contains C software to implement JPEG image encoding,
decoding, and transcoding. JPEG is a standardized compression method for
full-color and gray-scale images.
The distributed programs provide conversion between JPEG "JFIF" format and
image files in PBMPLUS PPM/PGM, GIF, BMP, and Targa file formats. The
core compression and decompression library can easily be reused in other
programs, such as image viewers. The package is highly portable C code; we
have tested it on many machines ranging from PCs to Crays.
We are releasing this software for both noncommercial and commercial use.
Companies are welcome to use it as the basis for JPEG-related products. We
do not ask a royalty, although we do ask for an acknowledgement in product
literature (see the README file in the distribution for details). We hope to
make this software industrial-quality --- although, as with anything that's free,
we offer no warranty and accept no liability.
For more information, contact jpeg-info@jpegclub.org.
Contents of this directory
jpegsrc.vN.tar.gz contains source code, documentation, and test files for
release N in Unix format.
jpegsrN.zip contains source code, documentation, and test files for release N
in Windows format.
D-9
jpegaltui.vN.tar.gz contains source code for an alternate user interface for

cjpeg/djpeg in Unix format.
jpegaltuiN.zip contains source code for an alternate user interface for
cjpeg/djpeg in Windows format.
wallace.ps.gz is a PostScript file of Greg Wallace's introductory article about
JPEG. This is an update of the article that appeared in the April 1991
Communications of the ACM.
jpeg.documents.gz tells where to obtain the JPEG standard and documents
about JPEG-related file formats.
jfif.ps.gz is a PostScript file of the JFIF (JPEG File Interchange Format) format
specification.
jfif.txt.gz is a plain text transcription of the JFIF specification; it's missing a
figure, so use the PostScript version if you can.
TIFFTechNote2.txt.gz is a draft of the proposed revisions to TIFF 6.0's JPEG
support.
pm.errata.gz is the errata list for the first printing of the textbook "JPEG Still
Image Data Compression Standard" by Pennebaker and Mitchell.
jdosaobj.zip contains pre-assembled object files for JMEMDOSA.ASM.
If you want to compile the IJG code for MS-DOS, but don't have an assembler,
these files may be helpful.
libMng
COPYRIGHT NOTICE:
Copyright © 2000-2008 Gerard Juyn (gerard@libmng.com)
For the purposes of this copyright and license, "Contributing Authors" is
defined as the following set of individuals:
Gerard Juyn
(hopefully some more to come...)
The MNG Library is supplied "AS IS". The Contributing Authors disclaim all
warranties, expressed or implied, including, without limitation, the
warranties of merchantability and of fitness for any purpose. The
Contributing Authors assume no liability for direct, indirect, incidental,
special, exemplary, or consequential damages, which may result from the use
of the MNG Library, even if advised of the possibility of such damage.
Permission is hereby granted to use, copy, modify, and distribute this source
code, or portions hereof, for any purpose, without fee, subject to the following
restrictions:
1. The origin of this source code must not be misrepresented.
2. Altered versions must be plainly marked as such and must not be
misrepresented as being the original source.
D - 10
3. This Copyright notice may not be removed or altered from any source or
altered source distribution.
The Contributing Authors specifically permit, without fee, and encourage the
use of this source code as a component to supporting the MNG and JNG file
format in commercial products. If you use this source code in a product,
acknowledgment would be highly appreciated.
libPng
COPYRIGHT NOTICE, DISCLAIMER, and LICENSE:
If you modify libpng you may insert additional notices immediately following
this sentence.
This code is released under the libpng license.
libpng versions 1.2.6, August 15, 2004, through 1.2.40, September 10, 2009, are
Copyright (c) 2004, 2006-2009 Glenn Randers-Pehrson, and are distributed
according to the same disclaimer and license as libpng-1.2.5 with the following
individual added to the list of Contributing Authors
Cosmin Truta
libpng versions 1.0.7, July 1, 2000, through 1.2.5 - October 3, 2002, are
Copyright (c) 2000-2002 Glenn Randers-Pehrson, and are distributed
according to the same disclaimer and license as libpng-1.0.6 with the following
individuals added to the list of Contributing Authors
Simon-Pierre Cadieux
Eric S. Raymond
Gilles Vollant
and with the following additions to the disclaimer:
There is no warranty against interference with your enjoyment of the library
or against infringement. There is no warranty that our efforts or the library
will fulfill any of your particular purposes or needs. This library is provided
with all faults, and the entire risk of satisfactory quality, performance,
accuracy, and effort is with the user.
libpng versions 0.97, January 1998, through 1.0.6, March 20, 2000, are
Copyright (c) 1998, 1999 Glenn Randers-Pehrson, and are distributed
according to the same disclaimer and license as libpng-0.96, with the following
individuals added to the list of Contributing Authors:
Tom Lane
Glenn Randers-Pehrson
Willem van Schaik
libpng versions 0.89, June 1996, through 0.96, May 1997, are Copyright (c) 1996,
1997 Andreas Dilger Distributed according to the same disclaimer and license
as libpng-0.88, with the following individuals added to the list of Contributing
Authors:
D - 11
John Bowler
Kevin Bracey
Sam Bushell
Magnus Holmgren
Greg Roelofs
Tom Tanner
libpng versions 0.5, May 1995, through 0.88, January 1996, are Copyright (c)
1995, 1996 Guy Eric Schalnat, Group 42, Inc.
For the purposes of this copyright and license, "Contributing Authors" is
defined as the following set of individuals:
Andreas Dilger
Dave Martindale
Guy Eric Schalnat
Paul Schmidt
Tim Wegner
The PNG Reference Library is supplied "AS IS". The Contributing Authors
and Group 42, Inc. disclaim all warranties, expressed or implied, including,
without limitation, the warranties of merchantability and of fitness for any
purpose. The Contributing Authors and Group 42, Inc. assume no liability for
direct, indirect, incidental, special, exemplary, or consequential damages,
which may result from the use of the PNG Reference Library, even if advised
of the possibility of such damage.
Permission is hereby granted to use, copy, modify, and distribute this source
code, or portions hereof, for any purpose, without fee, subject to the following
restrictions:
1. The origin of this source code must not be misrepresented.
2. Altered versions must be plainly marked as such and must not be
misrepresented as being the original source.
3. This Copyright notice may not be removed or altered from any source or
altered source distribution.
The Contributing Authors and Group 42, Inc. specifically permit, without fee,
and encourage the use of this source code as a component to supporting the
PNG file format in commercial products. If you use this source code in a
product, acknowledgment is not required but would be appreciated.
Libpng is OSI Certified Open Source Software. OSI Certified Open Source is a
certification mark of the Open Source Initiative.
Glenn Randers-Pehrson
glennrp at users.sourceforge.net
September 10, 2009
D - 12
libSRTP
Copyright (c) 2001-2005 Cisco Systems, Inc.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
* Neither the name of the Cisco Systems, Inc. nor the names of its
contributors may be used to endorse or promote products derived from this
software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND
CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES,
INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDERS OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
libssh2
Copyright (c) 2004-2007 Sara Golemon <sarag@libssh2.org>
Copyright (C) 2006-2007 The Written Word, Inc.
Redistributions of source code must retain the above copyright notice, this
Redistributions in binary form must reproduce the above copyright notice,
Neither the name of the copyright holder nor the names of any other
contributors may be used to endorse or promote products derived from this
software without specific prior written permission.
D - 13
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND

CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES,
INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
ncurses
Copyright (c) 1998-2004, 2006 Free Software Foundation, Inc.
copy, modify, merge, publish, distribute, distribute with modifications,
sublicense, and/or sell copies of the Software, and to permit persons to whom
the Software is furnished - to do so, subject to the following conditions:
PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE ABOVE
COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR
OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR
OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
Except as contained in this notice, the name(s) of the above copyright holders
shall not be used in advertising or otherwise to promote the sale, use or other
dealings in this Software without prior written authorization.
OpenLDAP
The OpenLDAP Public License
Version 2.8, 17 August 2003
Redistribution and use of this software and associated documentation
("Software"), with or without modification, are permitted provided that the
following conditions are met:
D - 14
1. Redistributions in source form must retain copyright statements and

notices,
2. Redistributions in binary form must reproduce applicable copyright
statements and notices, this list of conditions, and the following
disclaimer in the documentation and/or other materials provided with
the distribution, and
3. Redistributions must contain a verbatim copy of this document.
The OpenLDAP Foundation may revise this license from time to time.
Each revision is distinguished by a version number. You may use this
Software under terms of this license revision or under the terms of any
subsequent revision of the license.
THIS SOFTWARE IS PROVIDED BY THE OPENLDAP FOUNDATION AND
ITS CONTRIBUTORS `ÀS IS'' AND ANY EXPRESSED OR IMPLIED
WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
OPENLDAP FOUNDATION, ITS CONTRIBUTORS, OR THE AUTHOR(S)
OR OWNER(S) OF THE SOFTWARE BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY
WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
The names of the authors and copyright holders must not be used in
advertising or otherwise to promote the sale, use or other dealing in this
Software without specific, written prior permission. Title to copyright in this
Software shall at all times remain with copyright holders.
OpenLDAP is a registered trademark of the OpenLDAP Foundation.
Copyright 1999-2003 The OpenLDAP Foundation, Redwood City, California,
USA. All Rights Reserved. Permission to copy and distribute verbatim copies
of this document is granted.
OpenSSL
The OpenSSL toolkit stays under a dual license, i.e. both the conditions of the
OpenSSL License and the original SSLeay license apply to the toolkit. See
below for the actual license texts. Actually both licenses are BSD-style Open
Source licenses. In case of any license issues related to OpenSSL please contact
openssl-core@openssl.org.
OpenSSL License
Copyright (c) 1998-2008 The OpenSSL Project. All rights reserved.
D - 15
1. Redistributions of source code must retain the above copyright notice, this
2. Redistributions in binary form must reproduce the above copyright notice,
3. All advertising materials mentioning features or use of this software must
display the following acknowledgment:
"This product includes software developed by the OpenSSL Project for use in
the OpenSSL Toolkit. (http://www.openssl.org/)"
4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to
endorse or promote products derived from this software without prior written
permission. For written permission, please contact openssl-core@openssl.org.
5. Products derived from this software may not be called "OpenSSL" nor may
"OpenSSL" appear in their names without prior written permission of the
OpenSSL Project.
6. Redistributions of any form whatsoever must retain the following
acknowledgment:
"This product includes software developed by the OpenSSL Project for use in
the OpenSSL Toolkit (http://www.openssl.org/)"
THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT `ÀS IS'' AND
ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
EVENT SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER
IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
This product includes cryptographic software written by Eric Young
(eay@cryptsoft.com). This product includes software written by Tim Hudson
(tjh@cryptsoft.com).
Original SSLeay License:
Copyright (C) 1995-1998 Eric Young (eay@cryptsoft.com)
This package is an SSL implementation written by Eric Young
(eay@cryptsoft.com).
The implementation was written so as to conform with Netscape’s SSL.
This library is free for commercial and non-commercial use as long as the
following conditions are adhered to. The following conditions apply to all
code found in this distribution, be it the RC4, RSA, lhash, DES, etc., code; not
just the SSL code. The SSL documentation included with this distribution is
covered by the same copyright terms except that the holder is Tim Hudson
(tjh@cryptsoft.com).
Copyright remains Eric Young's, and as such any Copyright notices in the
code are not to be removed. If this package is used in a product, Eric Young
D - 16
should be given attribution as the author of the parts of the library used. This
can be in the form of a textual message at program startup or in documentation
(online or textual) provided with the package. Redistribution and use in
source and binary forms, with or without modification, are permitted
provided that the following conditions are met:
1. Redistributions of source code must retain the copyright notice, this list of
conditions and the following disclaimer.
3. All advertising materials mentioning features or use of this software must
display the following acknowledgement: "This product includes
cryptographic software written by Eric Young (eay@cryptsoft.com)"
The word 'cryptographic' can be left out if the routines from the library being
used are not cryptographic related.
4. If you include any Windows specific code (or a derivative thereof) from the
apps directory (application code) you must include an acknowledgement:
"This product includes software written by Tim Hudson (tjh@cryptsoft.com)"
THIS SOFTWARE IS PROVIDED BY ERIC YOUNG `ÀS IS'' AND ANY
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL
THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY
WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
The licence and distribution terms for any publicly available version or
derivative of this code cannot be changed. i.e. this code cannot simply be
copied and put under another distribution licence [including the GNU Public
Licence.]
pmap-29092002
Copyright (c) 2002 Andrew Isaacson <adi@hexapodia.org>
1. Redistributions of source code must retain the above copyright notice, this
D - 17
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR `ÀS IS'' AND ANY
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL
THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
zlib
version 1.2.3, July 18th, 2005
Copyright (C) 1995-2005 Jean-loup Gailly and Mark Adler
This software is provided 'as-is', without any express or implied warranty. In
no event will the authors be held liable for any damages arising from the use
of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it freely,
subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not claim
that you wrote the original software. If you use this software in a product, an
acknowledgment in the product documentation would be appreciated but is
not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
Jean-loup Gailly Mark Adler
jloup@gzip.org madler@alumni.caltech.edu
D - 18
Index
Numerics Ethernet call control A–72
802.1Q VLAN header 4–82 event logging A–104
feature A–116
A finder A–118
fonts A–92
access URL 4–68 gains A–50
ACD. See also automatic call distribution graphic icons A–103
acoustic echo cancellation 4–79 H.235 A–112
acoustic echo cancellation <aec> A–54 H.323 A–22
acoustic echo suppression <aes> A–55 hold, local reminder A–85
idle display A–120
AEC. See also acoustic echo cancellation
indicator classes A–102
AGC. See also automatic gain control indicator patterns A–102
alert information A–19 indicators, assignments A–102
animations <anim> A–101 IP TOS call control A–74
application configuration keep-alive A–79
acoustic echo cancellation A–54 keys A–94
acoustic echo suppression A–55 local camera view A–71
animations A–101 local protocol A–8
audio codec preferences A–45 localization A–27
audio codec profiles A–49 main browser A–121
automatic gain control A–57 multilingual A–28
background noise suppression A–56 music on hold A–21
backgrounds A–96 network monitoring A–75
bitmaps A–100 outbound proxy A–18
call handling configuration A–80 password lengths A–109
call progress patterns A–40 platform A–100
camera controls A–70 port A–78
chord-sets A–36 power saving A–132
compliance A–22 presence A–91
conference setup A–20 protocol A–7
connection reuse A–21 protocol server A–8
date and time A–31 protocol special events A–20
dial plan A–23 provisioning A–114
dial plan, emergency A–27 Quality of Service A–71
dialog A–21 RAM disk A–115
directory A–85 receive equalization A–57
DNS cache A–126 request A–115
dual tone multi-frequency A–35 request delay A–116
encryption A–109 request validation A–19
resource A–118
ring type A–43
routing server A–27
Index – 1
RTP A–72, A–73, A–78 basic protocols

sampled audio for sound effects A–37 header support B–4
SDP A–17 hold implementation B–9
security A–108 request support B–3
shared calls A–84 response support B–6
SIP A–11 RFC and Internet draft support B–2
soft keys A–128 transfer B–9
sound effect patterns A–39 basic TCP/IP A–75
sound effects A–38 behaviors <behaviors> A–153
tones A–34
blind transfers 4–20
transmit equalization A–59
user preferences A–31 BNS. See also background noise suppression
video A–64 boot failure messages 5–8
video codec preferences A–66 boot server security policy 3–16
video codec profiles A–66 boot servers
voice activity detection A–60 deploying phones 3–17
voice coding algorithms redundant 3–14
voice coding algorithms <codecs> A–44 security policy 3–16
voice settings A–44 setting up 3–15
volume persistence A–50
bootROM 2–3
web server A–79
bootROM and application wrapper 2–4
application configuration file A–5
bootROM error messages 5–2
application error messages 5–3
bootROM tasks 2–3
application files, overview 2–6
bootROM/SIP application dependencies C–9
Applications key 4–31
bridged line appearance signaling B–10
attendant <attendant> A–151
bridged line appearances 4–54
attended transfers 4–20
browser limits A–122
audio codec iLBC 4–80
busy <busy> A–143
audio codec preferences <codecPref> A–45
busy lamp field 4–55
audio codec profiles <audioProfile> A–49
audio codecs 4–80 C
audio playback feature 4–36, A–117
call auto answer A–83
audio recording feature 4–36, A–117
call auto routing A–84
automatic call distribution <acd> A–156 call control <callControl> A–72
automatic gain control 4–82
call control, third party B–9
automatic gain control <agc> A–57 call forwarding 4–22, A–142
automatic off-hook call placement 4–19 call handling configuration <call> A–80
automatic off-hook call placement
call hold 4–19
<autoOffHook> A–140
call log 4–3
B call park/retrieve 4–24
background logo call progress patterns A–40
adding C–6 call progress tones, synthesized 4–30
configuration file changes C–7 call timer 4–3
background noise suppression 4–81 call transfer 4–20
background noise suppression <ns> A–56 call waiting 4–3
backgrounds <bg> A–96 called party identification 4–4
basic logging A–106 calling party identification 4–4
calls <calls> A–139
camera controls <camera> A–70
Index – 2
Index
central provisioning, overview 2–6 distinctive incoming call treatment 4–7

changed features distinctive ringing 4–7
SIP 3.2 2–15 diversion A–142
changing the key on the phone C–6 DND. See also do not disturb
chord-sets <chord> A–36 DNS cache <dns> A–126
comfort noise fill 4–82 DNS SIP server name resolution 4–61
compliance <compliance> A–22 do not disturb 4–8
conference setup <conference> A–20 do not disturb <dnd> A–140, A–144
configurable feature keys 4–26 downloadable fonts 4–30
configurable soft keys 4–48 DTMF event RTP payload 4–79
configuration file encryption 4–95 DTMF tone generation 4–79
configuration file example 4–65 DTMF. See also dual tone multi-frequency
connected party identification 4–5 dual tone multi-frequency <DMTF> A–35
connection reuse <connectionReuse> A–21 dynamic noise reduction 4–84
consultative transfers 4–20
context sensitive volume control 4–5 E
corporate directory 4–34 electronic hookswitch, supported 4–9, A–155
corporate directory feature A–87, A–117 emergency <emergency> A–27, A–148
custom certificates 4–93 emergency routing A–27, A–148
customizable audio sound effects 4–6 encryption <encryption> A–109
customizable fonts and indicators 4–28 enhanced feature keys
definition language 4–39
D examples 4–46
date and time <datetime> A–31 macro definitions 4–42
useful tips 4–45
default feature key layouts C–13
enhanced feature keys feature 4–38, A–117
default password 3–6, 4–99, C–11, C–27
Ethernet IEEE 802.1p/Q A–71
deploying phones from the boot server 3–17
Ethernet menu 3–12
device <device> A–157
device certificates, support for 4–95 F
DHCP INFORM 3–3, 3–8, 3–9
feature <feature> A–116
DHCP menu 3–8
feature licensing 4–21, 4–22, 4–34, 4–36, 4–83, 4–
DHCP or manual TCP/IP setup 3–2 87, A–117
DHCP, secondary server 3–3 feature synchronized ACD feature 4–59, A–156
diagnostics, phone 5–10 features
dial plan <dialplan> A–23 list of 1–6
dialog <dialog> A–21 finder <finder> A–118
digit map flash parameter configuration A–157
default A–26 flash parameter. See also device
examples A–24
fonts <font> A–92
match and replace A–25
protocol A–25 forward all <fwd> A–143
timer A–25
G
digit map <digitmap> A–147
digital picture frame feature A–117 gains <gain> A–50
directed call pick-up 4–24 graphic display backgrounds 4–17, A–96
directory <dir> A–85 graphic icons <gi> A–103
distinctive call waiting 4–8 group call pick-up 4–24
Index – 3
H local reminder <localReminder> A–85

H.235 <H235> A–112 local user and administrator privilege levels 4–92
H.323 <H323> A–22 localization <lcl> A–27
H.323 protocol 2–15 log files 5–5
handset, headset, and speakerphone 4–9 logging <log> A–104
hands-free, disabled A–33 low-delay audio packet transmission 4–77
hold <hold> A–85
M
I MAC address
idle display <idleDisplay> A–120 definition A–2
substitution 3–17, 3–18, 3–25, A–4
idle display animation 4–16
main browser <main> A–121
iLBC 4–80
main menu 3–7
incoming signaling validation 4–93
manage conferences 4–22
indicator classes <class> A–102
manual configuration, overview 2–7
indicators A–100
manual log upload 5–7
indicators, assignments A–102
manual routing A–33
installing SIP application 3–17
master configuration files
instant messaging 4–29
details A–2
IP TOS A–72 model number version A–5
IP TOS call control <callControl> A–74 overview 2–5
IP_400 font A–94 part number substitution A–4
IP_500 font A–94 message waiting indication 4–7
IP_600 font A–94 message waiting indicator <mwi> A–149
messaging <msg> A–149
J Microbrowser 4–31, 4–68
jitter buffer 4–78 microphone mute 4–14
Microsoft Live Communications Server 2005
K Integration 4–64
keep-alive <keepalive> A–79 migration dependencies C–10
key features 1–6 miscellaneous patterns A–42
keys <key> A–94 missed call configuration <serverMissedCall>
A–141
L missed call notification 4–5
languages, adding new A–29 model number substitution A–5
languages, supported 4–29 modifying network configuration 3–6
last call return 4–25 multilingual <ml> A–28
LDAP directory, virtual list view support A–90 multilingual user interface 4–29
LEDs A–103 multiple call appearances 4–28
length <length> A–109 multiple line keys per registration 4–27
link layer discovery protocol C–30 multiple registrations 4–57
LLDP. See also link layer discovery protocol music on hold 4–20
local / centralized conferencing 4–21 music on hold <musicOnHold> A–21
local <local> A–8 mutual TLS, support for 4–97
local camera view <localCameraView> A–71
local contact directory 4–10
local contact directory file format 4–11
local digit map 4–13
Index – 4
Index
N Polycom HDX
Network Address Translation <nat> A–150 supported software C–10
network configuration, modifying 3–6 Polycom VVX 1500
power saving feature 4–52
network monitoring <netMon> A–75
Polycom VVX 1500 D
new features
H.323 protocol 2–15, 4–87
SIP 3.1.2 2–14
SIP 3.1.3 2–14 port <port> A–78
SIP 3.2 2–14 power saving <powerSaving> A–132
SIP 3.2.2 2–15 presence 4–64
no answer <noanswer> A–144 presence <pres> A–91
product-model-part number mapping C–26
O protocol <voIpProt> A–7
Option 66 3–8 protocol server <server> A–8
outbound proxy <outboundProxy> A–18 protocol special events <specialEvent> A–20
provisioning <prov> A–114
P
provisioning protocols 3–4
packet error concealment 4–78
provisioning protocols, supported 3–4
password <pwd> A–109
patterns <pat> A–39 Q
patterns <pattern> A–102 QOS. See also Quality of Service
peer networking <pnet>application Quality of Service <QOS> A–71
configuration
quick setup feature 4–76
peer networking A–125
quotas <quotas> A–119
per-phone configuration
attendant A–151
R
automatic call distribution A–156
automatic off-hook call placement A–140 RAM disk <ramdisk> A–115
behaviors A–153 rebooting phones 3–19, 3–23
busy A–143 receive equalization <rxEq> A–57
calls A–139 registration <reg> A–134
dial plan, emergency A–148
digit map A–147 reliability of provisional responses B–9
do not disturb A–140, A–144 request <request> A–115
forward all A–143 request delay <delay> A–116
message waiting indicator A–149 request validation <requestValidation> A–19
messaging A–149 resetting to factory defaults 3–6
missed call configuration A–141
resource <res> A–118
Network Address Translation A–150
no answer A–144 resource files, overview 2–8
quotas A–119 resource list <resourceList> A–152
registration A–134 RFC support B–2
resource list A–152 ring type <rt> A–43
roaming buddies A–154 ringer patterns A–41
roaming privacy A–154
roaming buddies <roaming_buddies> A–154
routing A–147
routing server A–148 roaming privacy <roaming_provacy> A–154
per-phone configuration file A–133 routing <routing> A–147
phone diagnostics 5–10 routing server <server> A–27, A–148
phone quick setup 4–76 RTP <RTP> A–72, A–73, A–78
phone1.cfg A–133
Index – 5
S SIP header
sampled audio files A–38 diversion A–14
warning A–15
sampled audio for sound effects <saf> A–37
SIP headers, warnings 4–75
SCA. See also shared call appearances
SIP. See also Session Initiation Protocol
scheduled logging parameters A–107
sip.cfg A–5
screen capture, phone A–33
SIP<SIP> A–11
SDP <SDP> A–17
SIP-B automatic call distribution 4–58
secure real-time transport protocol 4–93
soft keys <softkey> A–128
security <sec> A–108
sound effects <se> A–38
server menu 3–10
SoundPoint IP
server redundancy 4–60
applications 4–32
server-based call forwarding See also call configuring phones locally 4–98
forwarding device certificates 4–95
server-based DND See also do not disturb features, list of 1–6
Services key. See also Applications key supported languages 4–29
Session Initiation Protocol SoundPoint IP / SoundStation IP / VVX phones
setting up changed features, overview 2–15
advanced features 4–25 features, overview 2–8
audio features 4–77 introduction 1–1
basic features 4–1 network 2–2
boot server 3–14 new features, overview 2–14, 2–15
network 3–2 SoundPoint IP 32x/33x
security features 4–92 switching text entry mode 3–8
shared call appearance signaling B–10 SoundPoint IP 650
shared call appearances playback 4–36, A–117
shared calls <shared> A–84 recording 4–36, A–117
shared lines SoundPoint IP 670
barge-in 4–53, A–136 playback 4–36, A–117
recording 4–36, A–117
SIP
1xx Responses - Provisional B–6 SoundStation IP
2xx Responses - Success B–7 applications 4–32
3xx Responses - Redirection B–7 configuring phones locally 4–98
4xx Responses - Request Failure B–7 device certificates 4–95
5xx Responses - Server Failure B–8 features, list of 1–6
6xx Responses - Global Failure B–9 supported languages 4–29
application architecture 2–3 SoundStation IP 7000
basic protocols, hold implementation B–9 supported software C–10
basic protocols, request support B–3 treble/bass controls 4–84
basic protocols, response support B–6 speed dial 4–15
basic protocols, RFC and Internet draft SRTP. See also secure real-time transport
support B–2 protocol
basic protocols, transfer B–9 static DNS cache 4–72
instant messaging and presence leveraging
status menu 5–4
extensions B–10
RFC 2–1 supported LDAP directories 4–34
SIP application
description 2–4 T
installing 3–17 text entry mode, switching 3–8
upgrading 3–21 time and date display 4–15
SIP basic protocols, header support B–4 time synchronization A–75
Index – 6
Index
TLS. See also transport layer security voice activity detection <vad> A–60
TLVs. See also type length values voice mail integration 4–56
transmit equalization <txEq> A–59 voice quality monitoring 4–83, A–61
transport layer security voice setting <voice> A–44
troubleshooting volume persistence <volume> A–50
Application is not compatible 5–2
application error messages 5–3 W
application logging options 5–5 web server <httpd> A–79
audio issues 5–16
blinking time 5–4 welcome sound, reboot A–38
boot failure messages 5–8
bootROM error messages 5–2
calling issues 5–14
Config file error. Error is 5–3
controls issues 5–12
Could not contact boot server 5–2
displays issues 5–15
Error loading 5–3
Error, application is not present! 5–3
Failed to get boot parameters via DHCP 5–2
log files 5–5
manual log upload 5–7
Network link is down 5–3
Not all configuration files were present 5–3
power and startup issues 5–11
productivity suite 5–16
reading a boot log 5–8
reading an application log 5–9
registration status 5–4
scheduled logging 5–6
screens and systems access issues 5–13
trusted certificate authority list C–1
type length values
type-of-service bits 4–82
U
uaCSTA A–14, A–134, B–9
upgrading SIP application 3–21
USB device 4–36
USB devices, supported 4–36
user interface, soft key activated 4–14
user preferences <up> A–31
V
VAD. See also voice activity detection
video <video> A–64
video codec preferences <codecPref> A–66
video codec profiles <profile> A–66
VLAN ID using DHCP C–23
voice activity detection 4–78
Index – 7
Index – 8
POLYCOM, INC.
APPLICATION PROGRAMMING INTERFACE LICENSE (“API”)
FOR SOUNDPOINT IP AND SOUNDSTATION IP PRODUCTS (“Product” or “Products”).
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6. Harmful Code. You agree not to include any “Harmful Code” in any products you develop by use of the API, including but not limited
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programs without control of a person operating the computing equipment on which it resides, or (iii) retrieves or collects information
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8. No Warranty. You understand the API provided to you is supplied "AS IS" AND “WITH ALL FAULTS” WITHOUT ANY
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Spip Ssip VVX Admin Guide SIP 3 2 2 Eng

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Spip Ssip VVX Admin Guide SIP 3 2 2 Eng

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Administrator’s Guide for the

SIP 3.2.2 | November 2009 | 1725-11530-322 Rev. A

© 2009 Polycom, Inc. All rights reserved.

The Administrator’s Guide for the SoundPoint IP/SoundStation IP/VVX

• Quick Start Guides, which describe how to assemble the phones

• Developer’s Guide, which assists in the development of applications that

• Technical Bulletins, which describe workarounds to existing issues and

About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . iii

1 Introducing the SoundPoint IP / SoundStation IP / VVX

3 Setting up Your System . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–1

4 Configuring Your System . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1

Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29

Voice Quality Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–83

5 Troubleshooting Your SoundPoint IP / SoundStation IP / VVX

A Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .A–1

Diversion <divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–142

B Session Initiation Protocol (SIP) . . . . . . . . . . . . . . . . . . . . . B–1

C Miscellaneous Administrative Tasks . . . . . . . . . . . . . . . . . . C–1

Modifying Phone’s Configuration Using the Web Interface . . . . . . . . . C–27

D Third Party Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–1

This chapter introduces the SoundPoint IP / SoundStation IP / VVX family,

• SoundPoint IP Desktop Phones

• SoundStation IP Conference Phones

• Polycom VVX 1500 Business Media Phone

SoundPoint IP Desktop Phones

The currently supported desktop phones are:

SoundStation IP Conference Phones

The currently supported conference phones are:

Polycom VVX 1500 Business Media Phone

Key Features of Your SoundPoint IP / SoundStation IP / VVX

• Award winning sound quality and full-duplex speakerphone or

• Dual auto-sensing 10/100/1000baseT Ethernet ports

• Power over Ethernet (PoE) port or Power pack option

• Multiple language support on most phones

• Polycom Browser on the Polycom VVX 1500

• XML status/control API

This chapter provides an overview of the Session Initiation Protocol (SIP)

• A working IP network is established.

• Routers are configured for VoIP.

• VoIP gateways are configured for SIP.

• The latest (or compatible) SoundPoint IP / SoundStation IP / VVX phone

• Where SoundPoint IP / SoundStation IP / VVX Phones Fit

• Session Initiation Protocol Application Architecture

• New Features in SIP 3.2

refer to Configuring Your System on page 4-1. To troubleshoot any problems

Where SoundPoint IP / SoundStation IP / VVX Phones Fit

Session Initiation Protocol Application Architecture

• BootROM—loads first when the phone is powered on

• SIP Application—software that makes the device a phone

• Configuration—configuration parameters stored in separate files

• Resource Files—optional, needed by some of the advanced features

5. Downloads the master configuration file.

BootROM and SIP Application Wrapper

Warning Configuration files should only be modified by a knowledgeable system

The phone configuration files consist of:

• Master Configuration Files

• Application Configuration Files

Master Configuration Files

• Specified master configuration file

• Per-phone master configuration file

• Default master configuration file

Application Configuration Files